Best 111 Text to Speech Tools of 2025

Text to Bark is the first AI-powered text-to-speech model developed by ElevenLabs, designed to help people communicate more effectively with their dogs. This technology not only demonstrates high-quality speech synthesis but also simulates dog sounds naturally, creating a communication method suitable for dogs to understand. The launch of this innovative product elevates the interaction between humans and pets to a new level, making communication between owners and their dogs more interesting and effective. Users can generate corresponding "dog language" through simple text input, thereby better understanding and interacting with their pets.

This is a powerful text-to-speech generator with over 1000 high-quality AI voices. Suitable for various use cases such as podcasts, education, and business content creation. Users can leverage this platform to generate clear, natural-sounding voice content, supporting voice cloning and audio/video editing. Reasonably priced at only $39.99 per month, it's suitable for both individuals and businesses.

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.

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.

Spark-TTS is a highly efficient text-to-speech synthesis model based on large language models, featuring single-stream decoupled speech tokens. Leveraging the power of large language models, it directly reconstructs audio predicted from code, omitting the additional acoustic feature generation model, thus improving efficiency and reducing complexity. This model supports zero-shot text-to-speech synthesis, enabling cross-lingual and code-switching scenarios, making it ideal for speech synthesis applications requiring high naturalness and accuracy. It also supports virtual voice creation; users can generate different voices by adjusting parameters such as gender, pitch, and speaking rate. The model aims to address the inefficiencies and complexities of traditional speech synthesis systems, providing a highly efficient, flexible, and powerful solution for research and production. Currently, the model is primarily intended for academic research and legitimate applications such as personalized speech synthesis, assistive technologies, and language research.

Llasa is a text-to-speech (TTS) base model based on the Llama framework, designed for large-scale speech synthesis tasks. The model is trained using 160,000 hours of tokenized speech data and has efficient language generation capabilities and multilingual support. Its main advantages include powerful speech synthesis capabilities, low inference costs, and flexible framework compatibility. This model is suitable for education, entertainment, and commercial scenarios, providing users with high-quality speech synthesis solutions. This model is currently freely available on Hugging Face, aiming to promote the development and application of speech synthesis technology.

Octave TTS is a next-generation speech synthesis model developed by Hume AI. It not only converts text to speech but also understands the semantics and emotions of the text to generate expressive speech output. The core advantage of this technology lies in its deep understanding of language, allowing it to generate natural and vivid speech based on context. It is suitable for various application scenarios, including audiobooks, virtual assistants, and expressive voice interaction. The emergence of Octave TTS marks the development of speech synthesis technology from simple text reading to a more expressive and interactive direction, providing users with a more personalized and emotional voice experience. Currently, this product is primarily aimed at developers and creators, providing services through APIs and platforms. Future expansion to more languages and application scenarios is expected.

IndexTTS is a GPT-style text-to-speech (TTS) model primarily developed based on XTTS and Tortoise. It can correct Chinese pronunciation using pinyin and control pauses using punctuation marks. This system introduces a character-pinyin mixed modeling method in Chinese scenarios, significantly improving training stability, timbre similarity, and audio quality. Furthermore, it integrates BigVGAN2 to optimize audio quality. The model is trained on tens of thousands of hours of data and outperforms current popular TTS systems such as XTTS, CosyVoice2, and F5-TTS. IndexTTS is suitable for scenarios requiring high-quality speech synthesis, such as voice assistants and audiobooks, and its open-source nature makes it suitable for academic research and commercial applications.

ElevenReader Publishing, powered by ElevenLabs, is an innovative platform that leverages AI audio models to transform books into high-quality audiobooks. It solves the problems of high costs and complex processes associated with traditional audiobook production, offering authors a fast, free, and globally distributed solution. The platform supports multiple file formats, allows users to preview audio and select their preferred AI voice, and provides listener reports and analytics to help authors understand their audience. Its key advantages are zero cost, rapid generation, and global distribution, making it ideal for independent authors and publishers.

ElevenLabs Studio is a platform focused on audio content creation, leveraging advanced AI technology to convert text into high-quality audio. Key advantages include support for multiple file formats, a vast voice library, and the ability to adjust voice expression based on emotion and context. Suitable for audiobook production and podcast creation, it helps creators efficiently generate audio content, improving both efficiency and quality. Pricing may vary depending on user needs and usage; refer to the official website for pricing details.

NVIDIA's PDF to Podcast Blueprint is a generative AI-based application that can transform PDF documents (such as training materials, technical studies, or documentation) into personalized audio content. This technology leverages large language models (LLMs), text-to-speech (TTS) technology, and NVIDIA NIM microservices to convert PDF data into engaging audio content, facilitating learning on the move while addressing information overload. The solution runs entirely on NVIDIA's cloud infrastructure, eliminating the need for local GPU hardware, ensuring compliance with privacy regulations, and offering customization options for branding, analytics, real-time translation, or digital avatar interfaces based on user needs.

Zonos is an advanced text-to-speech model that supports multiple languages and can generate natural speech based on text prompts along with speaker embeddings or audio prefixes. It also features voice cloning, allowing for accurate replication of a speaker's voice with just a few seconds of reference audio. The model delivers high-quality speech output (44kHz) and allows fine control over speech rate, pitch variation, audio quality, and emotional tone (such as happiness, fear, sadness, and anger). Zonos offers Python and Gradio interfaces for easy user onboarding and supports deployment through Docker. The model achieves a real-time factor of approximately 2 times on an RTX 4090, making it suitable for applications that require high-quality speech synthesis.

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.

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.

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.

Hailuo AI Audio leverages advanced speech synthesis technology to convert text into natural and fluent speech. Its main advantages include generating high-quality and expressive speech suitable for various scenarios such as audiobook production and voice broadcasting. The product is positioned as a professional-grade audio synthesis tool that currently offers a limited-time free trial, aiming to provide users with an efficient and convenient voice generation solution.

kokoro-onnx is a text-to-speech (TTS) project based on the Kokoro model and ONNX runtime. It supports English and plans to support French, Japanese, Korean, and Chinese. The model offers near real-time performance on macOS M1 and provides a variety of voice options, including whispering. The model is lightweight, approximately 300MB (around 80MB when quantized). This project is open-source on GitHub under the MIT license, facilitating easy integration and use for developers.

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.

TangoFlux is an efficient text-to-audio (TTA) generation model with 515M parameters, capable of generating up to 30 seconds of 44.1kHz audio in just 3.7 seconds on a single A40 GPU. The model introduces the CLAP-Ranked Preference Optimization (CRPO) framework to address the alignment challenges of TTA models, enhancing TTA alignment through iterative generation and optimization of preference data. TangoFlux achieves state-of-the-art performance in both objective and subjective benchmark tests, and all code and models are open-source to support further research in TTA generation.

Nijivoice is a voice generation platform leveraging artificial intelligence technology, allowing users to generate emotionally resonant voices by selecting different characters and inputting text. The significance of this technology lies in its ability to provide personalized voices that meet a variety of needs, from entertainment to business, while being easy to operate and user-friendly. Background information about the product shows that Nijivoice offers various voice options suitable for different scenarios, including VTubers, virtual characters, corporate introduction videos, product promotions, and educational content. In terms of pricing, Nijivoice offers a free plan as well as various paid plans to accommodate different user needs.

The ChatGPT Podcast Generator is a platform that leverages artificial intelligence to help users quickly convert text content into podcast episodes. With AI voices, an audio editor, and collaborative features, it enables content creators, marketers, and individuals with stories to tell to easily produce high-quality podcast content. This product meets the demand for audio content in a fast-paced digital media environment, thanks to its user-friendliness, efficiency, and lack of need for professional recording equipment.

CosyVoice Speech Generation Model 2.0-0.5B is a high-performance speech synthesis model that supports zero-shot and cross-language synthesis, enabling direct generation of speech output based on text content. Offered by Tongyi Laboratory, it boasts powerful speech synthesis capabilities and a wide range of applications, including but not limited to intelligent assistants, audiobooks, and virtual hosts. The model's significance lies in its ability to provide natural and fluent speech output, greatly enhancing the experience of human-machine interaction.

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.

AI Podcast Generator is an online service that rapidly converts PDF files and webpage content into high-quality audio formats using professional AI voices and customizable speaking styles, ensuring perfect content delivery. The significance of this technology lies in its substantial enhancement of content accessibility and diversity, allowing information to be disseminated quickly in audio format, particularly beneficial for users needing to convert text into audio for various scenarios. Product background information indicates it offers swift processing, high-quality output, and enterprise-level solutions, with various subscription plans available to cater to diverse user needs.

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.

PlayNote is a product that leverages cutting-edge AI voice synthesis technology to convert various files and data into audio creations. It supports multiple file formats, including PDFs, CSVs, TXTs, as well as image formats like PNG, JPEG, video formats like MP4, MOV, and audio formats like WAV, MP3. Users can upload files, and PlayNote will convert the content into audio, enabling users to listen in various situations. The significance of this technology lies in its ability to enhance information accessibility, especially for visually impaired individuals or users who need to obtain information without reading. Background information shows that it is provided by PlayAI, aiming to improve work efficiency and quality of life through technological innovation. For pricing, users can visit the Pricing page for more details.

AI Voice Lab is a free AI text-to-speech tool that leverages cutting-edge GPT-like AI voice model technology to deliver highly realistic voice results. It supports over 20 languages and 100+ voice styles, providing free usage opportunities daily and suitable for various scenarios such as video and audio production, enhancing content attraction.

Read To Me is an online service that enables users to convert PDF files into audio format for listening on various devices, enhancing the accessibility and efficiency of information retrieval. Key advantages of this technology include one-click conversion, the ability to listen anytime and anywhere, productivity enhancement, straightforward pricing, clear audio quality, and secure file handling. The background information indicates that Read To Me is designed to reduce the need for prolonged screen time, allowing users to learn while commuting, exercising, or doing household chores. In terms of pricing, Read To Me operates on a per-file payment basis with no hidden fees or recurring subscription charges.