Best 54 Speech Recognition Tools of 2025

parakeet-tdt-0.6b-v2 is a 600 million parameter automatic speech recognition (ASR) model designed to achieve high-quality English transcription with accurate timestamp prediction and automatic punctuation and capitalization support. The model is based on the FastConformer architecture, capable of efficiently processing audio clips up to 24 minutes long, making it suitable for developers, researchers, and various industry applications.

Kimi-Audio is an advanced open-source audio foundation model designed to handle a variety of audio processing tasks, such as speech recognition and audio dialogue. The model has been extensively pre-trained on over 13 million hours of diverse audio and text data, giving it strong audio reasoning and language understanding capabilities. Its key advantages include excellent performance and flexibility, making it suitable for researchers and developers to conduct audio-related research and development.

Amazon Nova Sonic is a cutting-edge foundational model that integrates speech understanding and generation, enhancing the natural fluency of human-computer dialogue. This model overcomes the complexities of traditional voice applications, achieving a deeper level of communication understanding through a unified architecture. It is suitable for AI applications across multiple industries and holds significant commercial value. As AI technology continues to develop, Nova Sonic will provide customers with better voice interaction experiences and improved service efficiency.

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.

DuRT is a speech recognition and translation tool focusing on macOS. It uses local AI models and system services to achieve real-time speech recognition and translation, supporting multiple speech recognition methods to improve accuracy and language support. The product displays results in a floating window for easy access during use. Its main advantages include high accuracy, privacy protection (no user information is collected), and a convenient user experience. DuRT is positioned as a highly efficient productivity tool, designed to help users communicate and work more efficiently in multilingual environments. The product is currently available on the Mac App Store; pricing is not explicitly mentioned on the page.

Scribe is a high-accuracy speech-to-text model developed by ElevenLabs, designed to handle the unpredictability of real-world audio. It supports 99 languages and provides features such as word-level timestamps, speaker diarization, and audio event labeling. Scribe demonstrates superior performance on the FLEURS and Common Voice benchmarks, surpassing leading models like Gemini 2.0 Flash, Whisper Large V3, and Deepgram Nova-3. It significantly reduces error rates for traditionally underserved languages (such as Serbian, Cantonese, and Malayalam), where error rates often exceed 40% in competing models. Scribe offers an API for developer integration and will launch a low-latency version to support real-time applications.

Supertone Play is a platform dedicated to voice cloning and AI voice content creation. Leveraging advanced AI technology, it empowers users to create personalized voice content through simple voice inputs. This technology has wide-ranging applications in entertainment, education, business, and more, providing users with a novel means of expression and creation. The platform's voice cloning feature allows users to rapidly create unique voice models, while AI voice content creation generates high-quality voice content based on user requirements. The key advantages of this technology are its efficiency, personalization, and innovative nature, catering to diverse user needs in voice creation.

Step-Audio is the first production-level open-source intelligent voice interaction framework, integrating voice understanding and generation capabilities. It supports multilingual dialogue, emotional intonation, dialects, speech rate, and prosodic style control. Its core technologies include a 130B parameter multimodal model, a generative data engine, fine-grained voice control, and enhanced intelligence. This framework promotes the development of intelligent voice interaction technology through open-source models and tools, and is suitable for a variety of voice application scenarios.

FireRedASR-AED-L is an open-source, industrial-grade automatic speech recognition model designed to meet the needs for high efficiency and performance in speech recognition. This model utilizes an attention-based encoder-decoder architecture and supports multiple languages including Mandarin, Chinese dialects, and English. It achieved new record levels in public Mandarin speech recognition benchmarks and has shown exceptional performance in singing lyric recognition. Key advantages of the model include high performance, low latency, and broad applicability across various speech interaction scenarios. Its open-source feature allows developers the freedom to use and modify the code, further advancing the development of speech recognition technology.

FireRedASR is an open-source industrial-grade Mandarin automatic speech recognition model, utilizing an Encoder-Decoder and LLM integrated architecture. It includes two variants: FireRedASR-LLM and FireRedASR-AED, designed for high-performance and efficient needs respectively. The model excels in Mandarin benchmarking tests and also performs well in recognizing dialects and English speech. It is suitable for industrial applications requiring efficient speech-to-text conversion, such as smart assistants and video subtitle generation. The open-source model is easy for developers to integrate and optimize.

PengChengStarling is an open-source toolkit focused on multilingual automatic speech recognition (ASR), developed based on the icefall project. It supports the entire ASR process, including data processing, model training, inference, fine-tuning, and deployment. By optimizing parameter configurations and integrating language identifiers into the RNN-Transducer architecture, it significantly enhances the performance of multilingual ASR systems. Its main advantages include efficient multilingual support, a flexible configuration design, and robust inference performance. The models in PengChengStarling perform exceptionally well across various languages, require relatively small model sizes, and offer extremely fast inference speeds, making it suitable for scenarios that demand efficient speech recognition.

Whisper Turbo is a speech recognition tool optimized based on the Whisper Large-v3 model, specifically designed for fast voice transcription. It leverages advanced AI technology to efficiently convert speech from various audio sources into text, supporting multiple languages and accents. This tool is offered to users at no cost, aiming to help people save time and energy and enhance their productivity. It primarily serves users who require quick and accurate voice content transcription, such as bloggers, content creators, and businesses, providing them with a convenient speech-to-text solution.

RealtimeSTT is an open-source speech recognition model capable of converting spoken language into text in real time. It employs advanced voice activity detection technology to automatically detect the start and end of speech without manual intervention. Additionally, it supports wake word activation, allowing users to initiate speech recognition by saying specific wake words. The model is characterized by low latency and high efficiency, making it suitable for real-time transcription applications such as voice assistants and meeting notes. It is developed in Python, easy to integrate and use, and is open-source on GitHub, with an active community that continuously provides updates and improvements.

MinMo, developed by Alibaba Group's Tongyi Laboratory, is a multimodal large language model with approximately 8 billion parameters, focused on achieving seamless voice interactions. It is trained on 1.4 million hours of diverse voice data through various stages, including speech-to-text alignment, text-to-speech alignment, speech-to-speech alignment, and full-duplex interaction alignment. MinMo achieves state-of-the-art performance across various benchmarks in speech understanding and generation, while maintaining the capabilities of text-based large language models and supporting full-duplex dialogues, enabling simultaneous bidirectional communication between users and the system. Additionally, MinMo introduces a novel and straightforward voice decoder that surpasses previous models in speech generation. Its command-following ability has been enhanced to support voice generation control based on user instructions, including details such as emotion, dialect, and speech rate, as well as mimicking specific voices. MinMo's speech-to-text latency is approximately 100 milliseconds, with theoretical full-duplex latency around 600 milliseconds, and actual latency around 800 milliseconds. The development of MinMo aims to overcome the major limitations of previous multimodal models, providing users with a more natural, smooth, and human-like voice interaction experience.

BetterWhisperX is an improved automatic speech recognition model based on WhisperX, offering fast speech-to-text services with word-level timestamps and speaker identification features. This tool is vital for researchers and developers handling large volumes of audio data, as it significantly enhances the efficiency and accuracy of speech data processing. The product is built on OpenAI's Whisper model with further optimizations and improvements. Currently, the project is free and open-source, aiming to provide the developer community with more efficient and accurate speech recognition tools.

The LiveKit Plugins Turn Detector is a plugin designed for LiveKit Agents that leverages a custom open-weight model to determine when a user has finished speaking, thereby introducing end-to-end speech endpoint detection. Compared to conventional Voice Activity Detection (VAD) models, this plugin utilizes a language model specifically trained for this task, offering a more accurate and robust method for detecting speech endpoints. Currently, it only supports English and is not recommended for other languages.

BoldVoice Accent Oracle is an online tool that can quickly identify the accent of users when they speak English and guess their native language. The significance of this technology lies in its ability to help language learners understand their pronunciation traits, facilitating targeted improvements. BoldVoice's background information shows that the company is committed to enhancing communication skills through technology, and the tool may be used in educational and language learning contexts. The website does not provide specific pricing information, but given its educational nature, it may offer free trials or basic services at no charge, with advanced features available for a fee.

Moonshine Web is a simple application built with React and Vite, running on Moonshine Base, a powerful speech recognition model optimized for fast and accurate automatic speech recognition (ASR), particularly suited for resource-constrained devices. The application runs locally in the browser, utilizing Transformers.js and WebGPU for acceleration (with WASM as an alternative). Its significance lies in providing users with a serverless solution for local speech recognition, which is especially crucial for scenarios requiring swift processing of voice data.

OmniAudio-2.6B is a multimodal model with 2.6 billion parameters that seamlessly processes both text and audio inputs. This model combines Gemma-2B, Whisper Turbo, and a custom projection module. Unlike the traditional method of chaining ASR and LLM models, it unifies both capabilities in an efficient architecture, achieving minimal latency and resource overhead. This enables it to securely and rapidly process audio-text directly on edge devices such as smartphones, laptops, and robots.

Voice Control is a product launched by Hume AI that employs an interpretative approach for AI voice customization. It allows developers to finely tune AI voices by continuously adjusting 10 voice dimensions (such as gender, assertiveness, vibrancy, etc.) without relying on voice cloning technology. This method not only enhances the precision of voice customization but also ensures the reproducibility of voice modifications across different sessions. The launch of Voice Control signifies a major advancement in AI voice customization technology, enabling developers to easily tailor the perfect voice for their brands or applications through an intuitive no-code interface.

Transcribro is a private, on-device speech recognition keyboard and text service application for the Android platform. It utilizes whisper.cpp to run OpenAI's Whisper series model and integrates Silero VAD for voice activity detection. The app provides a speech input keyboard, allowing users to input text via speech, and can be explicitly used by other applications or set as the user's preferred speech-to-text app. Some applications may use it for speech-to-text conversion. Transcribro's foundation is to offer users a safer and more private speech-to-text solution, avoiding potential privacy breaches associated with cloud processing. The application is open source, enabling users to freely view, modify, and distribute the code.

Universal-2 is the latest speech recognition model launched by AssemblyAI, surpassing the previous Universal-1 in both accuracy and precision. It captures the complexities of human language more effectively, providing users with audio data that requires no secondary verification. The significance of this technology lies in its ability to deliver sharper insights, faster workflows, and an exceptional product experience. Universal-2 features notable improvements in proper noun recognition, text formatting, and alphanumeric recognition, consequently reducing word error rates in practical applications.

Voice Changer is an audio modulation model launched by Cartesia that allows users to transform audio voices while maintaining the original expression and emotion. This technology is based on Cartesia's groundbreaking work in state-space model (SSM) architecture, capable of processing and generating high-resolution sound with astonishing quality. Key advantages of Voice Changer include the preservation of natural speech, precise control over delivery, diverse usage scenarios, and integration with Sonic voice generation technology.

Moonshine is a suite of speech-to-text models optimized for resource-constrained devices, making it ideal for real-time, on-device applications such as live transcription and voice command recognition. It outperforms the OpenAI Whisper model of the same size in word error rate (WER) on test datasets used in the OpenASR leaderboard maintained by HuggingFace. Additionally, Moonshine's computational requirements vary with the length of the input audio, allowing for quicker processing of shorter audio compared to the Whisper model, which processes everything in 30-second chunks. Moonshine processes 10-second audio segments at a speed five times faster than Whisper while maintaining the same or better WER.

GLM-4-Voice is an end-to-end voice model developed by a team from Tsinghua University, capable of directly understanding and generating Chinese and English speech for real-time dialogue. Leveraging advanced speech recognition and synthesis technologies, it achieves seamless conversion from speech to text and back to speech, boasting low latency and high conversational intelligence. The model is optimized for intellectual engagement and expressive synthesis capabilities in the voice modality, making it suitable for scenarios requiring real-time voice interaction.

ElevenLabs Voice Design is an online platform that allows users to design and generate custom voices through simple text prompts. The significance of this technology lies in its ability to quickly create voices that match specific descriptions, such as age, accent, tone, or character, including fictional characters like trolls, elves, and aliens. It provides a powerful tool for audio content creators, advertisers, game developers, and others for various commercial and creative projects. ElevenLabs offers a free trial for users to register and try out their services.

Whispo is a speech dictation tool that leverages artificial intelligence technology to convert users' speech into text in real-time. Utilizing OpenAI's Whisper technology for voice recognition, it supports custom API use for transcription and allows for post-processing with large language models. Whispo is compatible with various operating systems, including macOS (Apple Silicon) and Windows x64, and ensures user privacy by storing all data locally. It is designed to improve the efficiency of users who require significant text input, whether for programming, writing, or everyday note-taking. Whispo is currently available for free trial, although specific pricing strategies have not been clearly stated on the website.

Flow by Wispr is an application dedicated to improving voice input efficiency. Utilizing advanced speech recognition technology, it allows users to type three times faster than traditional keyboard methods. Flow is particularly suited for those who need to quickly capture and edit text, such as writers, journalists, students, and professionals. Currently, the product only supports Mac computers with Apple Silicon, with plans for expansion to more platforms in the future.

Llama3-s v0.2 is a multimodal checkpoint developed by Homebrew Computer Company, focusing on improving speech comprehension capabilities. This model enhances its performance through early integration of semantic tagging and community feedback to streamline its structure, improve compression efficiency, and ensure consistent feature extraction from speech. Llama3-s v0.2 demonstrates stable performance across multiple speech understanding benchmarks and offers a live demo for users to experience its functionalities firsthand. Although the model is still in early development and has certain limitations—such as sensitivity to audio compression and a maximum handling time of 10 seconds for audio—the team intends to address these issues in future updates.

Audio Chat is a dedicated website for processing audio files, allowing users to upload audio recordings from lectures, meetings, or interviews for dialogue analysis. Utilizing advanced audio processing technology, this product helps users quickly grasp key points of the conversation, enhancing learning and work efficiency.