# Text-to-Speech

Chatterbox is the first open-source production-grade text-to-speech (TTS) model released by Resemble AI, featuring outstanding performance and stability. It shows superior results compared to top closed-source systems. The unique aspect of this model is its support for exaggerated emotional control, making it ideal for use in video games, AI agents, and various other scenarios. Chatterbox offers strong price competitiveness and supports super-low latency, making it suitable for production use.

Dia is a text-to-speech (TTS) model developed by Nari Labs, featuring 160 million parameters, capable of generating highly realistic conversations directly from text. The model supports emotion and intonation control and can generate non-verbal communication such as laughter and coughs. Its pre-trained model weights are hosted on Hugging Face and are suitable for English generation. This product is crucial for research and educational purposes, enabling advancements in conversational AI technology.

MegaTTS 3 is a highly efficient speech synthesis model based on PyTorch, developed by ByteDance, with ultra-high-quality speech cloning capabilities. Its lightweight architecture contains only 0.45B parameters, supports Chinese, English, and code switching, and can generate natural and fluent speech from input text. It is widely used in academic research and technological development.

OpenAI.fm is an interactive demonstration platform that allows developers to experience the latest text-to-speech models gpt-4o-transcribe, gpt-4o-mini-transcribe, and gpt-4o-mini-tts in the OpenAI API. This technology generates natural and fluent speech, making text content vivid and easy to understand. It is suitable for a variety of application scenarios, especially in voice assistants and content creation, helping developers better communicate with users and enhance user experience. This product is positioned for efficient speech synthesis and is suitable for developers who wish to integrate speech functions.

Orpheus TTS is an open-source text-to-speech system based on the Llama-3b model, aiming to provide more natural human speech synthesis. It boasts strong voice cloning and emotional expression capabilities, suitable for various real-time applications. This product is free and aims to provide developers and researchers with a convenient speech synthesis tool.

CSM 1B is a speech generation model based on the Llama architecture, capable of generating RVQ audio codes from text and audio input. The model is primarily used in speech synthesis and boasts high-quality speech generation capabilities. Its advantages include the ability to handle multi-speaker dialogue scenarios and generate natural and fluent speech through contextual information. This open-source model is intended to support research and educational purposes but is explicitly prohibited from being used for impersonation, fraud, or illegal activities.

Zonos TTS is an advanced AI text-to-speech technology supporting multiple languages, emotion control, and zero-shot voice cloning. It generates natural, expressive speech suitable for various scenarios, including education, audiobooks, video games, and voice assistants. The technology provides users with an efficient and personalized speech generation solution through high-quality audio output (44kHz) and fast real-time processing capabilities. While not entirely free, it offers flexible pricing plans to meet the needs of different users.

Kokoro TTS is a powerful text-to-speech tool that supports multiple languages and voice blending features, capable of converting EPUB, PDF, and TXT files into high-quality speech output. The tool provides developers and users with flexible voice customization options to easily create professional-grade audio. Its main advantages include multilingual support, voice blending, flexible input formats, and a free commercial license. This product is positioned to provide creators, developers, and businesses with an efficient and low-cost speech synthesis solution, suitable for audiobook creation, video narration, podcast production, educational content generation, and customer service, among other scenarios.

Lemonfox.ai Text-to-Speech API is an API service focusing on text-to-speech (TTS). It leverages advanced AI technology to quickly convert text into natural and fluent speech, supporting multiple languages and accents, suitable for various scenarios such as voice broadcasting and audiobook production. Its main advantages include low cost, high quality, and easy integration, enabling enterprises or developers to quickly implement voice functions and enhance user experience. This product is positioned as an efficient and cost-effective TTS solution for enterprises and developers, offering reasonable pricing, a free trial, and high value for money.

Developed by Zyphra, Zonos-v0.1-hybrid is an open-source text-to-speech model capable of generating highly natural speech based on text prompts. The model is trained on extensive English voice data, employing eSpeak for text normalization and phoneme processing, and predicting DAC tokens via a transformer or hybrid backbone network. It supports multiple languages, including English, Japanese, Chinese, French, and German, and allows for fine control over speech speed, pitch, audio quality, and emotion. Additionally, it features zero-shot voice cloning, requiring only 5 to 30 seconds of speech samples to achieve high-fidelity voice replication. The model operates with a real-time factor of about 2x on an RTX 4090, offering fast performance. It is equipped with an easy-to-use gradio interface and can be easily installed and deployed using Docker. Currently, the model is available on Hugging Face for free, but users need to deploy it themselves.

Zonos-v0.1 is a real-time text-to-speech (TTS) model developed by the Zyphra team, equipped with high-fidelity voice cloning features. This model includes a 1.6B parameter transformer model and a 1.6B parameter hybrid model, both released under the Apache 2.0 open source license. It can generate natural and expressive speech from text prompts and supports multiple languages. Additionally, Zonos-v0.1 enables high-quality voice cloning from 5 to 30-second voice clips and can be adjusted based on speaking speed, pitch, quality, and emotion. Its key advantages include high generation quality, support for real-time interaction, and flexible voice control capabilities. The release of this model aims to advance research and development in TTS technology.

TurboTTS is an advanced AI-based text-to-speech tool. It can quickly convert written text into natural, realistic speech, supporting up to 70 languages and over 300 authentic voice types. The main advantages of this technology include its high-quality voice output, user-friendly interface, and rapid content generation capabilities. Background information shows that the platform has been used by over 228,000 creators worldwide, processing more than 50 million voiceover texts daily, with a 99.9% uptime guarantee and 98% user satisfaction rate. TurboTTS offers both free and paid plans suitable for personal and professional users.

Sonofa is an AI-based product that can convert various forms of reading content (such as webpages, PDF files, and text from images) into audio content in podcast form. This technology leverages advanced text-to-speech (TTS) and natural language processing (NLP) capabilities to convert text content into natural and fluent speech, enabling users to acquire information without reading. The main advantage of this product is its significant enhancement of flexibility and efficiency in information retrieval, especially for those who are unable to read during commuting, exercising, or leisure activities. Information about Sonofa suggests it aims to help users make better use of fragmented time through innovative approaches, thereby improving personal learning and work efficiency. Currently, the services offered by Sonofa may be subscription-based; specific pricing and positioning have not yet been clarified.

Orate is a powerful AI voice toolkit that can convert text into realistic speech and vice versa. It supports multiple mainstream AI service providers and offers the main advantage of a unified API, making it easy for developers to quickly integrate and use. This toolkit is suitable for application development requiring voice interaction features, such as smart voice assistants and voice broadcasting systems. Pricing and specific positioning are not yet clear, but based on its features and community feedback, it shows high practicality and developmental value.

Kokoro TTS is an AI model focused on text-to-speech conversion, primarily designed to transform text into natural, fluent voice output. Based on the StyleTTS 2 architecture with 82 million parameters, it delivers high-quality speech synthesis while maintaining efficient performance and low resource consumption. Its multilingual support and customizable voice packs cater to diverse user needs in various contexts, such as creating audiobooks, podcasts, and training videos, making it especially beneficial in the education sector by enhancing content accessibility and engagement. Furthermore, Kokoro TTS is open-source and free to use, providing significant cost-effectiveness.

Llasa-1B is a text-to-speech model developed by the Audio Lab at the Hong Kong University of Science and Technology. Based on the LLaMA architecture and integrated with speech tokens from the XCodec2 codec, it converts text into natural and fluent speech. The model has been trained on 250,000 hours of Chinese and English speech data and supports generating speech from plain text, as well as utilizing given voice prompts for synthesis. Its main advantage is the ability to produce high-quality multilingual speech, making it suitable for various applications such as audiobooks and voice assistants. The model is licensed under CC BY-NC-ND 4.0, prohibiting commercial use.

Llasa-3B is a powerful text-to-speech (TTS) model developed based on the LLaMA architecture, focused on Chinese and English speech synthesis. By integrating XCodec2's speech encoding technology, it efficiently converts text into natural and fluent speech. Its main advantages include high-quality speech output, support for multilingual synthesis, and flexible speech prompting capabilities. This model is suitable for various applications requiring speech synthesis, such as audiobook production and voice assistant development. Its open-source nature also allows developers to explore and expand its functionalities freely.

Kokoro-82M is a text-to-speech (TTS) model created by hexgrad and hosted on Hugging Face. It features 82 million parameters and is open-sourced under the Apache 2.0 license. The model released version 0.19 on December 25, 2024, offering 10 unique voice packages. Kokoro-82M ranks first in the TTS Spaces Arena, showcasing its efficiency in parameter scale and data usage. It supports both American and British English, making it suitable for generating high-quality speech output.

opensource_notebooklm is an open-source project designed to achieve natural and educational dialogue generation by combining Deepseek-V3 language understanding and PlayHT text-to-speech technology. The project can generate podcast-like dialogues, making it suitable for both education and entertainment. Its main advantages include powerful language generation capabilities and high-quality voice output, providing significant value in educational content creation and language learning applications.

Voice Cursor is an experimental text editor built on the native audio capabilities of Gemini 2.0, demonstrating how to integrate Gemini's new text-to-speech API into a text editor for smooth and contextual voice generation. This project not only showcases the powerful new features of Gemini 2.0 but also provides a practical application example, allowing developers and users to explore and utilize this new technology. The product background includes innovative projects from Google Creative Lab aimed at pushing technological boundaries and providing new modes of interaction. Currently, the product is free, primarily targeting developers and technology enthusiasts seeking innovative solutions to enhance productivity and improve accessibility.

Paper-to-Podcast is a tool that converts academic papers into podcast format, simulating a discussion among three individuals to help listeners comprehend the paper's content in a more natural and humanized way. It not only makes complex information easier to digest but also provides valuable insights and critical analysis. The tool employs the OpenAI API for text-to-speech conversion, generating realistic voices with distinct character traits, allowing listeners to absorb the content of research papers by listening rather than reading while commuting or traveling.

ElevenLabs Conversational AI is a voice agent product that can be rapidly deployed on websites, mobile devices, or phones. It features low latency, full configurability, and seamless scalability, supporting turn-taking and interruption handling in natural conversations, making it suitable for unpredictable dialogues in noisy environments. The product combines speech-to-text, large language models (LLM), and text-to-speech technologies, supporting multiple languages and customizable voices for various scenarios including customer support, scheduling, and outbound sales.

Auralis is a text-to-speech (TTS) engine that converts text into natural speech quickly, supports voice cloning, and boasts extremely fast processing speeds—capable of handling an entire novel in just minutes. The product is distinguished by its high speed, efficiency, easy integration, and high-quality audio output, making it suitable for scenarios requiring rapid text-to-speech conversion. Built on a Python API, Auralis supports long text streaming, built-in audio enhancement, automated language detection, and more. Developed by AstraMind AI, Auralis aims to provide a practical TTS solution for real-world applications. While product pricing is not explicitly stated on the page, the codebase is released under the Apache 2.0 License, allowing for free use in projects.

ElevenReader is an application that utilizes AI technology to convert text content, such as PDFs, articles, and e-books, into podcasts. It generates intelligent podcasts, enabling users to listen to content anytime, anywhere. According to product background information, ElevenLabs aims to help users consume and experience content in new ways through high-quality AI audio technology. GenFM on ElevenReader supports multiple languages to meet the needs of global users.

OuteTTS-0.2-500M is a text-to-speech synthesis model built on Qwen-2.5-0.5B. It has been trained on a larger dataset, achieving significant improvements in accuracy, naturalness, vocabulary range, voice cloning capability, and multilingual support. Special thanks to Hugging Face for the GPU funding that supported this model's training.

ElevenLabs Projects is a platform focused on producing long-form audio content, allowing users to transform books and scripts into audiobooks and podcasts. The product supports various file formats, features an extensive voice library, and offers AI voice technology with emotional nuance and contextual adaptation. It also includes a range of advanced features such as multilingual support, specific text segment voice assignments, and segment editing. With its high-quality AI audio technology, ElevenLabs Projects helps creators and businesses share their stories globally.

OuteTTS is an experimental text-to-speech model that generates speech using pure language modeling techniques. Its significance lies in harnessing advanced language modeling technology to transform text into natural-sounding speech, which is crucial for applications like speech synthesis, voice assistants, and automated dubbing. Developed by OuteAI, it supports both Hugging Face and GGUF models and offers advanced features such as voice cloning through the interface.

Lightning is the latest text-to-speech model developed by smallest.ai, breaking barriers in performance and size in multimodal AI with its ultra-fast processing speed and compact footprint. The model supports various accents in languages like English and Hindi, with rapid plans to expand to more languages. Lightning's non-autoregressive architecture allows the simultaneous synthesis of entire audio clips, unlike traditional autoregressive models that require sequential audio generation. Key advantages of Lightning include high generation speed, small model size, multilingual support, and quick adaptation to new data. Background information indicates that the launch of Lightning aims to significantly reduce latency and costs for voicebot companies by streamlining their architectures. Pricing for Lightning starts at $0.04 per minute, offering customized pricing plans for enterprise customers using over 100,000 minutes monthly.

Fish Speech is a product focused on voice synthesis, utilizing advanced deep learning techniques to convert text into natural and fluent speech. The product supports multiple languages, including Chinese and English, and is suitable for scenarios requiring text-to-speech conversion, such as voice assistants and audiobook production. Fish Speech stands out for its high-quality voice output, ease of use, and flexibility. Additionally, background information indicates that the product is continuously updated with increased dataset sizes and improved quantizer parameters to provide better service.

Fish Agent V0.1 3B is a groundbreaking speech-to-speech model capable of capturing and generating environmental audio information with unprecedented accuracy. The model utilizes a non-semantic tagging architecture, eliminating the need for traditional semantic encoders/decoders. Additionally, it is a cutting-edge text-to-speech (TTS) model trained on 700,000 hours of multilingual audio content. As a continuation of the Qwen-2.5-3B-Instruct pre-trained version, it has been trained on 200 billion speech and text tags. The model supports eight languages, including English and Chinese, with approximately 300,000 hours of training data for each of these languages and around 20,000 hours for others.