Alternatives to MAI-Transcribe-2-Streaming

Compare MAI-Transcribe-2-Streaming alternatives for your business or organization using the curated list below. SourceForge ranks the best alternatives to MAI-Transcribe-2-Streaming in 2026. Compare features, ratings, user reviews, pricing, and more from MAI-Transcribe-2-Streaming competitors and alternatives in order to make an informed decision for your business.

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    Speechmatics

    Speechmatics

    Speechmatics

    Best-in-Market Speech-to-Text & Voice AI for Enterprises. Speechmatics delivers industry-leading Speech-to-Text and Voice AI for enterprises needing unrivaled accuracy, security, and flexibility. Our enterprise-grade APIs provide real-time and batch transcription with exceptional precision—across the widest range of languages, dialects, and accents. Powered by Foundational Speech Technology, Speechmatics supports mission-critical voice applications in media, contact centers, finance, healthcare, and more. With on-prem, cloud, and hybrid deployment, businesses maintain full control over data security while unlocking voice insights. Trusted by global leaders, Speechmatics is the top choice for best-in-class transcription and voice intelligence. 🔹 Unmatched Accuracy – Superior transcription across languages & accents 🔹 Flexible Deployment – Cloud, on-prem, and hybrid 🔹 Enterprise-Grade Security – Full data control 🔹 Real-Time & Batch Processing – Scalable transcription
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    Gemini 3.5 Transcribe
    Gemini 3.5 Transcribe is Google’s most precise speech-to-text model yet, designed for intelligent voice interactions and real-time transcription. Instead of simply converting speech word for word, it turns raw audio into accurate, polished, formatted text while handling background noise, complex jargon, accents, dialects, and natural speaking patterns. Smart transcription automatically understands self-corrections, removes filler words such as “ums” and “ahs,” and formats the final text for readability. The model supports continuous bidirectional streaming with sub-second latency for interactive voice applications, as well as pre-recorded audio processing for meetings, call logs, and other recordings with speaker attribution and word-level timestamps. Custom vocabulary helps it recognize specialized terminology, unique spellings, postal codes, order IDs, and other domain-specific language.
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    Cartesia Ink 2
    Ink 2 is Cartesia’s fastest, most accurate streaming speech-to-text model, built for production voice agents with the lowest word error rate and best turn detection of any streaming STT. It is designed to transcribe structured data such as phone numbers, dates, and emails correctly the first time, while also knowing when a speaker starts and finishes without requiring a separate voice activity detection system. Turn detection is built directly into the model, so voice agents can react to events instead of managing raw transcript segments. Ink 2 emits a full lifecycle of turn events, giving an agent clear signals for when to listen, interrupt, think, prepare a reply, cancel a premature response, or speak. The transcript property is cumulative within a turn, meaning each update contains the full text transcribed so far rather than a delta, and emitted text is final once sent.
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    Muse Voice Transcribe
    Muse Voice Transcribe is Meta’s first real-time audio perception model, delivering streaming automatic speech recognition (ASR), diarization, and endpointing in real time. An autoregressive multimodal model from the Muse Spark family, it processes audio in 80 ms chunks and decides dynamically whether to continue listening or emit text. Its adaptive delay changes the amount of audio context used for each word based on difficulty, balancing transcription accuracy with latency. The model is trained on more than 70 languages, with 25 extensively verified at launch, and natively supports arbitrary code-switching both within and between sentences. Language, keyword, and context biasing can further improve recognition accuracy for specific names, places, contacts, or terminology. Streaming diarization identifies speaker changes and distinguishes more than 20 speakers, while endpointing detects when speech begins and when a user finishes speaking.
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    Grok Speech to Text (STT)
    Grok Speech to Text is a standalone audio API built to help developers integrate fast, accurate transcription into any application. Built on the same stack that powers Grok Voice, Tesla vehicles, and Starlink customer support, the API is designed for use cases such as voice agents, real-time transcription tools, accessibility solutions, podcasts, meeting capture, telephony, and interactive audio experiences. Grok STT can generate transcripts from large audio files through a REST API or transcribe speech in real time through a low-latency WebSocket API. It includes word-level timestamps, speaker diarization, multichannel support, and intelligent Inverse Text Normalization that converts spoken language into properly formatted structured output for numbers, dates, currencies, and more. Grok Speech to Text is evaluated across phone calls, meetings, video and podcast content, and telephony, with strong performance in entity recognition and business use cases.
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    Voxtral Transcribe 2
    Voxtral Transcribe 2 is a next-generation family of speech-to-text models from Mistral AI that delivers ultra-low-latency, high-quality audio transcription and speaker diarization with broad language support. The suite includes Voxtral Mini Transcribe V2, optimized for batch transcription with features such as word-level timestamps, context biasing, and support for 13 languages, and Voxtral Realtime, designed specifically for live, streaming speech recognition with latency configurable down to sub-200 ms for real-time applications. Both models achieve state-of-the-art transcription accuracy while running efficiently and economically, with Mini Transcribe V2 offering leading performance and low error rates, and Realtime available as open source under the Apache 2.0 license so developers can deploy it on edge devices or in private environments.
    Starting Price: $14.99 per month
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    GPT‑Realtime‑Whisper
    GPT-Realtime-Whisper is OpenAI’s streaming transcription model built for low-latency speech-to-text experiences in live products. It transcribes audio as people speak, helping voice-enabled apps feel faster, more responsive, and more natural, from captions that appear in the moment to meeting notes that keep up with the conversation. It makes live speech usable inside business workflows as it happens, so teams can power captions for meetings, classrooms, broadcasts, and events, generate notes and summaries while conversations are still in progress, build voice agents that need to understand users continuously, and create faster follow-up workflows for high-volume spoken interactions. It is part of a new generation of real-time voice models in the API that can reason, translate, and transcribe as people speak, moving real-time audio beyond simple call-and-response toward voice interfaces that can listen, translate, transcribe, and take action as a conversation unfolds.
    Starting Price: $0.017 per minute
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    Azure AI Speech
    Build voice-enabled apps confidently and quickly with the Speech SDK. Transcribe speech to text with high accuracy, produce natural-sounding text-to-speech voices, translate spoken audio, and use speaker recognition during conversations. Create custom models tailored to your app with Speech studio. Get state-of-the-art speech to text, lifelike text to speech, and award-winning speaker recognition. Your data stays yours, your speech input is not logged during processing. Create custom voices, add specific words to your base vocabulary, or build your own models. Run Speech anywhere, in the cloud or at the edge in containers. Quickly and accurately transcribe audio in more than 92 languages and variants. Gain customer insights with call center transcription, improve experiences with voice-enabled assistants, capture key discussions in meetings and more. Use text to speech to create apps and services that speak conversationally, choosing from more than 215 voices, and 60 languages.
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    OpenAI Whisper
    Whisper is an automatic speech recognition (ASR) system developed by OpenAI for converting spoken language into text. It is trained on 680,000 hours of multilingual and multitask audio data collected from the web. The model is designed to handle diverse accents, background noise, and technical language with high accuracy. Whisper supports transcription in multiple languages as well as translation into English. It uses an encoder-decoder Transformer architecture to process audio inputs and generate text outputs. The system can also perform tasks like language identification and timestamp generation. Overall, Whisper enables developers to build robust voice-enabled applications with ease.
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    Cartesia Ink-Whisper
    Cartesia Ink is a family of real-time streaming speech-to-text (STT) models designed to power fast, natural conversations in voice AI applications, acting as the “voice input” layer that converts spoken language into accurate text instantly. Its flagship model, Ink-Whisper, is specifically engineered for conversational environments, delivering ultra-low latency transcription with a time-to-complete-transcript as fast as 66 milliseconds, enabling fluid, human-like interactions without noticeable delays. Unlike traditional transcription systems built for batch processing, Ink is optimized for live dialogue, handling fragmented, variable-length audio through dynamic chunking, which reduces errors and improves responsiveness during pauses, interruptions, or rapid exchanges.
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    NVIDIA Parakeet
    NVIDIA Parakeet-RNNT-1.1B is a multilingual automatic speech recognition model built for quality transcription across voice applications. With 1.1 billion parameters and training on more than 90,000 hours of speech, it supports 25 languages and regional variants, including English, Spanish, French, German, Italian, Arabic, Japanese, Korean, Portuguese, Russian, Hindi, Dutch, Danish, Norwegian, Czech, Polish, Swedish, Thai, Turkish, and Hebrew. The model automatically detects the spoken language and uses a universal tokenizer created by training language-specific tokenizers and merging them into a shared vocabulary, enabling efficient cross-lingual learning and deployment. Parakeet-RNNT produces case-sensitive transcripts with upper and lowercase text, punctuation, spaces, and apostrophes, making the output suitable for production voice applications and downstream language understanding.
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    MAI-Transcribe-1.5
    MAI-Transcribe-1.5 is Microsoft AI’s production-ready speech-to-text model for turning noisy audio into highly accurate, domain-aware transcripts across 43 languages. It delivers consistent, high-accuracy transcription across languages, accents, speaking styles, and challenging audio conditions, with automatic language detection included. The model is designed for real-world audio where speech often comes through conference rooms, phone lines, busy streets, low-quality recordings, background noise, and overlapping speakers. MAI-Transcribe-1.5 adapts transcription to domain-specific terminology, making it ready for captions, call analysis, accessibility, meeting transcription, doctor’s notes, pharma customer calls, content workflows, and other enterprise speech use cases out of the box. It uses contextual biasing to improve recognition of specialized vocabulary, names, industry language, and terms that generic transcription systems may miss.
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    MAI-Transcribe-1

    MAI-Transcribe-1

    Microsoft AI

    MAI-Transcribe-1 is a state-of-the-art speech-to-text model developed by Microsoft and available through Azure AI Foundry, designed to deliver high-accuracy transcription for real-world audio across enterprise and developer use cases. It supports 25 major languages and is optimized to handle diverse accents, dialects, and speaking styles, maintaining consistent performance even in challenging conditions such as background noise, low-quality recordings, or overlapping speech. It is built by Microsoft’s AI Superintelligence team with a dual focus on accuracy and efficiency, enabling fast batch transcription and scalable deployment for production environments. MAI-Transcribe-1 powers a wide range of applications, including meeting transcription, live captions, accessibility tools, call center analytics, and voice-driven agents, making it a foundational component for voice-enabled systems.
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    Scribe

    Scribe

    ElevenLabs

    ElevenLabs has introduced Scribe, an advanced Automatic Speech Recognition (ASR) model designed to deliver highly accurate transcriptions across 99 languages. Scribe is engineered to handle diverse real-world audio scenarios, providing features such as word-level timestamps, speaker diarization, and audio-event tagging. Benchmark tests, including FLEURS and Common Voice, demonstrate Scribe's superior performance over leading models like Gemini 2.0 Flash, Whisper Large V3, and Deepgram Nova-3, achieving the lowest word error rates in languages such as Italian (98.7%) and English (96.7%). Notably, Scribe also significantly reduces errors in languages that have been traditionally underserved, including Serbian, Cantonese, and Malayalam, where other models often exhibit error rates exceeding 40%. Developers can integrate Scribe through ElevenLabs' speech-to-text API, receiving structured JSON transcripts that include detailed annotations.
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    AccurateScribe.ai

    AccurateScribe.ai

    AccurateScribe.ai

    AccurateScribe.ai – AI-Powered Speech-to-Text Transcription for 134+ Languages. AccurateScribe.ai is an advanced, cloud-based speech-to-text transcription platform designed to deliver high-accuracy, multilingual voice transcription using cutting-edge AI models such as Whisper. With support for over 130 languages and dialects, the platform enables users to convert audio and video into precise, readable text—quickly and securely. Users can upload individual audio or video files in popular formats like MP3, WAV, MP4, and MOV, with support for files up to 10 hours or 5 GB in size. For added flexibility, AccurateScribe also offers an in-browser voice recorder that lets users record meetings, lectures, or notes directly and convert them into transcripts in real time. Additionally, users can transcribe public links from platforms such as YouTube, Dropbox, and Google Drive by simply pasting the URL—no manual downloads required.
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    Unmixr

    Unmixr

    Unmixr

    ​Unmixr is an AI-powered platform offering a suite of tools designed to enhance content creation and communication. Its text-to-speech feature supports over 1,300 human-like voices across 104 languages, allowing for the conversion of up to 200,000 characters of text into speech in a single request. The speech-to-text functionality provides accurate transcription of audio and video files, complete with speaker diarization and timestamping. For multilingual content, Unmixr's Dubbing Studio facilitates the translation and dubbing of audio and video into more than 100 languages through a streamlined process of transcription, translation, and dubbing. The AI chatbot integrates multiple models, including GPT-4o, Claude-3.5, Gemini Pro, and LLaMa-3.1, enabling users to engage in conversations and interact with documents such as PDFs and web pages. Additionally, Unmixr offers an AI image generator capable of producing high-quality images from text prompts, supporting various styles.
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    SpeechTexter

    SpeechTexter

    SpeechTexter

    SpeechTexter is a free multilingual speech-to-text application aimed at assisting you with transcription of any type of documents, books, reports or blog posts by using your voice. SpeechTexter allows adding custom voice commands for punctuation marks and some actions (undo, redo, make a new paragraph). Accuracy levels higher than 90% should be expected. It varies depending on the language and the speaker. SpeechTexter is used daily by students, teachers, writers, bloggers around the world. Voice-to-text software is exceptionally valuable for people who have difficulty using their hands due to trauma, people with dyslexia or disabilities that limit the use of conventional input devices. It will assist you in minimizing your writing efforts significantly. It can also be used as a tool for learning a proper pronunciation of words in the foreign language, in addition to helping a person develop fluency with their speaking skills. No download, installation or registration is required.
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    Voxtral

    Voxtral

    Mistral AI

    Voxtral models are frontier open source speech‑understanding systems available in two sizes—a 24 B variant for production‑scale applications and a 3 B variant for local and edge deployments, both released under the Apache 2.0 license. They combine high‑accuracy transcription with native semantic understanding, supporting long‑form context (up to 32 K tokens), built‑in Q&A and structured summarization, automatic language detection across major languages, and direct function‑calling to trigger backend workflows from voice. Retaining the text capabilities of their Mistral Small 3.1 backbone, Voxtral handles audio up to 30 minutes for transcription or 40 minutes for understanding and outperforms leading open source and proprietary models on benchmarks such as LibriSpeech, Mozilla Common Voice, and FLEURS. Accessible via download on Hugging Face, API endpoint, or private on‑premises deployment, Voxtral also offers domain‑specific fine‑tuning and advanced enterprise features.
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    Inworld Realtime STT
    Inworld Realtime STT is a realtime streaming STT API that understands users beyond their words. It combines low-latency speech recognition with voice profiling, extracting emotion, vocal style, accent, age, and pitch directly from raw audio so downstream LLMs and TTS systems can respond with more adaptive, expressive behavior. Developers can stream audio in real time, transcribe complete files, or extract voice profile signals through one unified API, with realtime bidirectional streaming over WebSocket, synchronous transcription for full audio files, voice profile signals on every streaming chunk, and multi-provider support through a single model ID. Every audio chunk can produce a realtime profile of the speaker with confidence scores, giving LLMs structured context such as whether a user sounds sad, frustrated, soft, high-pitched, or calm.
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    Supavocal

    Supavocal

    Supavocal

    Supavocal is an AI voice studio for text-to-speech, voice cloning, and speech-to-text. Turn text into expressive, studio-quality speech, clone any voice from about 15 seconds of audio, and transcribe recordings to text. Teams use Supavocal for video voiceovers, audiobook narration, character voices for games and animation, conversational chatbots and voice agents, and transcription, with a voice API for developers.
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    Spokenly

    Spokenly

    Spokenly

    Spokenly is an AI-powered dictation app for Mac, iPhone, Windows, and Linux that turns speech into clean, punctuated text wherever you work. Hold a shortcut, speak naturally, and release to place the transcription directly at the cursor in browsers, email, chat, word processors, IDEs, terminals, and other apps. It supports more than 100 languages, including mixed-language dictation, and offers both local and cloud speech-to-text models. Whisper, Parakeet, and other on-device models can run completely offline, while cloud engines from providers such as OpenAI, Deepgram, Groq, Soniox, and ElevenLabs can be used for higher-accuracy or real-time transcription. Local Only Mode blocks network requests so voice data stays on the device. Modes let users save different transcription models, AI providers, prompts, and output styles for specific tasks, while AI Instructions can remove filler words, fix grammar and punctuation, summarize, rewrite, translate, or reformat dictated text.
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    SpeechText.AI

    SpeechText.AI

    SpeechText.AI

    Transcribe audio and video into text. Get accurate transcriptions of podcasts with domain-specific speech recognition. SpeechText.AI is a powerful artificial intelligence software for speech to text conversion and audio transcription. Upload audio or video files. AI transcription software supports various file formats and transcribes from speech to text in any language. Select domain. Select industry domain and audio type from predefined categories to improve the recognition accuracy of domain-specific words. Transcribe. Our speech transcription engine uses state-of-the-art deep neural network models to convert from audio to text with close to human accuracy. Edit & Export. Search, modify and verify audio transcriptions using interactive editing tools. Export your content in different formats. Why SpeechText.AI? Set of amazing features to help you transcribe audio and video in seconds. Speech recognition. Powerful speech-to-text tech.
    Starting Price: $19 one-time payment
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    Rev AI
    Rev AI is a speech-to-text API platform that helps developers convert prerecorded audio and real-time audio streams into accurate transcripts. The platform is designed for accuracy, speed, global scale, and developer-friendly implementation. Rev AI supports speech-to-text across 57+ languages with grammar, punctuation, formatting, and low word error rates. Its proprietary models are trained using a large library of human-verified speech data to improve precision across voices, accents, and use cases. Rev AI also includes AI insights such as language identification, sentiment analysis, topic extraction, summarization, translation, and precise word-level timestamps. Built for developers and enterprises, Rev AI helps teams turn speech into searchable, analyzable, and actionable text.
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    Azure Speech to Text
    Quickly and accurately transcribe audio to text in more than 85 languages and variants. Customize models to enhance accuracy for domain-specific terminology. Get more value from spoken audio by enabling search or analytics on transcribed text or facilitating action, all in your preferred programming language. Get accurate audio to text transcriptions with state-of-the-art speech recognition. Add specific words to your base vocabulary or build your own speech-to-text models. Run Speech to Text anywhere, in the cloud or at the edge in containers. Access the same robust technology that powers speech recognition across Microsoft products. Convert audio to text from a range of sources, including microphones, audio files, and blob storage. Use speaker diarisation to determine who said what and when. Get readable transcripts with automatic formatting and punctuation. Tailor your speech models to understand organization- and industry-specific terminology.
    Starting Price: $1 per audio hour
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    Rekam AI

    Rekam AI

    Rekam AI

    Rekam AI is an all-in-one voice creation platform offering text to speech, speech to text, voice cloning, and AI voice generation. It uses high-quality, human-like voice models to transform written text into natural-sounding audio. Rekam AI provides a free text-to-speech tool that allows users to generate lifelike narration instantly. The platform includes a curated voice library with multiple male and female voices across accents and tones. Voice cloning enables users to create realistic digital voice replicas using short audio samples. Rekam AI also supports accurate speech-to-text transcription for meetings, interviews, and content creation. Overall, it serves as a complete voice studio for modern audio production.
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    SpokenData

    SpokenData

    ReplayWell

    Let the automatic speech-to-text technology transcribe your data. Or transcribe your data yourself or buy professional transcript. Use our on-line time synchonous editor to surf your data and transcripts. Download transcripts in many formats. Manage your team of transcribers using tags and categories. Help them with transcription by automatic voice-to-text technology. Integrate SpokenData into your application via our REST API. We adapt the voice-to-text on your data domain to maximize the transcript accuracy and lower your labor costs. Enable speech technologies in your applications through integrating SpokenData using our REST API. We are ready to process huge amounts of your data. You get API fitting your needs. Just contact our support team. We customize the voice-to-text on your data and purpose to maximize the transcript accuracy. Suitable for: web/mobile app developers, media monitoring agencies, audio/video archive business.
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    SpeechFlow

    SpeechFlow

    SpeechFlow

    SpeechFlow is a cutting-edge speech-to-text tool that empowers businesses and individuals with unparalleled accuracy and efficiency. Our advanced AI technology ensures precise transcription of audio and video content into written text, supporting up to 14 languages, beyond just English. Main Features: 1. Multilingual Transcriptions: Overcome language barriers with support for 14 languages. Get accurate and reliable transcriptions in diverse linguistic contexts. 2. All-in-One Transcription Solution: API & Online Platform:For enterprises and individuals, SpeechFlow offers a speech recognition API interface and online transcription features, which are simple and easy to use. 3. Accurate Transcriptions: Benefit from industry-leading accuracy, understanding industry-specific terminology, and context for comprehensive and reliable transcriptions.
    Starting Price: $0.0002 per second
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    Kukarella

    Kukarella

    Kukarella

    Kukarella is an AI-powered audio and voice-content platform that enables users to create professional voice-overs, multi-speaker dialogues, transcriptions, and visual content all within one integrated environment. The platform features a text-to-speech tool with access to hundreds of natural-sounding AI voices in more than 130 languages and accents, enabling rapid generation of voice narration without traditional recording studios or voice actors. It also supports audio transcription of uploads and online videos, extraction of text from webpages and images, voice-cloning for personalized narration, and a dialogue-generation tool that creates scripted conversations with distinct AI voices assigned automatically. In addition, users can translate and dub content into multiple languages, generate matching images or videos to complement their audio, and streamline workflows for e-learning, corporate narration, IVR voice-over, and multilingual content production.
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    Voice to Text Pro
    Redesigned from the ground up, Voice to Text Pro is the best tool for converting any audio into text. With Voice to Text Pro you won't need to type anything anymore, you just speak and your speech is instantly converted into text. It's also possible to transcribe audio from other sources files. Convert your speech to text, convert external files to text, share results to any app installed on your device or copy it to your clipboard, create notes based on your transcriptions or append text to existing notes. Sync your notes across all your devices, optimized support for iOS 14, iPhone 12, iPhone 12 Pro and iPads, and much more. Add frequently used words and expressions to increase transcription accuracy. Quick access to selected languages based on your preferences. Ad sponsors help us keep offering the free version. Becoming Premium you won't see ads anymore. With longer recordings, you are no longer limited to transcribe only 60 seconds of content at a time.
    Starting Price: $5.99 one-time payment
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    Beey

    Beey

    NEWTON Technologies

    Beey is an application which transcribes audio or video recordings into text with great accuracy in a few minutes. Beey can recognize speech in 20 languages. The user-friendly editor provides further processing of the transcribed text, export to various formats, and creating automatic subtitles or translation. The editor includes a recording preview synchronized with the edited text, which is illustrated by the moving cursor position. Editor controls allow slowing down, speeding up the playback, or starting the playback from the selected cursor position. Beey offers several additional tools: Link, Splitter, Stream and Voice. Link allows transcribing the video/audio directly from global platforms, such as YouTube. Splitter is convenient for working with long content. It splits the original recording into shorter ones, and users can work with them separately. Stream can perform real-time transcription, and caption ongoing streams. Voice records and transcribes live speech.
    Starting Price: €7.50 EUR per hour
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    Palatine Speech
    Palatine Speech is a cloud platform and API provider for AI-powered speech processing. It supports transcription, speaker diarization, word timestamps, automatic language detection, translation, SRT/VTT subtitles, sentiment analysis, and text summarization. The API supports streaming and asynchronous processing, custom dictionaries, OpenAI-compatible endpoints, more than 100 languages, and over 23 audio and video formats. Cloud and on-premise deployment are available. Palatine also develops Palatine Murmur 0.4.0, a privacy-first meeting recording, transcription, and AI-summary application available for macOS, Windows, and Linux.
    Starting Price: 0.29 RUB per audio minute
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    Neurotechnology AI SDK

    Neurotechnology AI SDK

    Neurotechnology

    Neurotechnology AI SDK is a multilingual toolkit for creating speech-to-text and voice processing applications. It combines a proprietary ASR engine for accurate transcription with a Speaker Diarization engine that separates and labels individual speakers in an audio stream. Supporting English, Lithuanian, Latvian and Estonian, it delivers fast performance on CPUs and GPUs for real-time or batch processing. Designed for on-premises use, all audio is processed locally, ensuring full data privacy and control. Its modular architecture lets developers use each component independently or integrate them into stand-alone or client-server systems. Optional speaker recognition through voice biometrics can be added for stronger identity confirmation. The SDK supports Windows and Linux and provides native libraries for Python, C++, Java and .NET, making it suitable for transcription workflows, analytics platforms or voice-driven applications across a wide range of industries.
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    GPT-Realtime-Translate
    GPT-Realtime-Translate is OpenAI’s live translation model for building multilingual voice experiences where each person can speak in their preferred language, hear the conversation translated in real time, and read real-time transcriptions. It supports more than 70 input languages and 13 output languages, making it useful for customer support, cross-border sales, education, events, media, and creator platforms serving global audiences. It is designed to preserve meaning while keeping pace with the speaker, even when people speak naturally, switch context, use regional pronunciation, or rely on domain-specific language. GPT-Realtime-Translate helps cross-language conversations feel more natural by combining lower latency, stronger fluency, and real-time speech translation in one API workflow. It can support live multilingual voice interactions, translate conversations as they happen, and make spoken content accessible to audiences.
    Starting Price: $0.034 per minute
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    EVI 3

    EVI 3

    Hume AI

    Hume AI's EVI 3 is a third-generation speech-language model that streams in user speech and forms natural, expressive speech and language responses. At conversational latency, it produces the same quality of speech as our text-to-speech model, Octave. Simultaneously, it responds with the same intelligence as the most advanced LLMs of similar latency. It also communicates with reasoning models and web search systems as it speaks, “thinking fast and slow” to match the intelligence of any frontier AI system. EVI 3 can instantly generate new voices and personalities instead of being limited to a handful of speakers. For instance, users can speak to any of the more than 100,000 custom voices already created on our text-to-speech platform, each with an inferred personality. No matter the voice, it responds with a wide range of emotions or styles, implicitly or on command.
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    StarWhisper

    StarWhisper

    StarWhisper

    StarWhisper is free voice-to-text software for Windows that lets you dictate anywhere with AI-powered transcription. It works offline with local Whisper AI or connects to OpenAI for 99% accuracy. Features include 29+ languages, GPU acceleration, wake word activation, auto-paste, file transcription, and multiple AI models. A free tier (500 words/day) covers casual use, while Pro plans unlock unlimited transcription and all models. Key Features: - Offline transcription with local Whisper AI - GPU acceleration for fast processing - 29+ language support - Wake word activation - Auto-paste into any app - File transcription - Multiple AI model sizes - OpenAI API integration Use Cases: - Dictate documents and emails - Transcribe meeting recordings - Voice-driven coding and notes - Accessibility for users with mobility issues - Multi-language content creation
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    Superwhisper

    Superwhisper

    Superwhisper

    Superwhisper is a voice-to-text app that lets users dictate, transcribe, and control writing workflows across apps without relying on the keyboard. The platform works on Mac, Windows, and iOS, with voice input that can be used in tools such as Slack, Cursor, Notion, Claude Code, Codex, and other agentic coding apps. Superwhisper supports push-to-talk, custom shortcuts, file transcription, meeting recording, custom modes, vocabulary settings, and AI-enhanced output. Users can create modes for different tasks, languages, tones, formats, prompts, and applications. The platform supports more than 100 languages and lets users choose from models such as GPT, Claude, Llama, Grok, Gemini, and others. Built for fast-moving professionals, developers, writers, and teams, Superwhisper helps turn speech into polished text, commands, transcripts, and AI-ready prompts.
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    EaseText Audio to Text Converter
    An intelligent tool to transcribe & convert audio to text freely. EaseText Audio to Text Converter is an offline AI-based automatic audio transcription software that uses artificial intelligence technology to transcribe & convert audio to text in real-time. The transcription can run offline on your computer to keep your data safe and secure. It supports a wide range of languages and offers high accuracy and a range of customization features, including the ability to transcribe multiple speakers and generate summaries of meetings and conversations. What's more, EaseText Audio to Text Converter supports saving the transcript file as TXT, WORD, HTML, PDF, etc. Features: 1 Convert audio file to text in high quality 2 Transcribe speech to text in real time 3 Record Meeting & take notes from Microsoft Teams, Google Meet, and Zoom 3 Enjoy high-speed batch file conversion 4 Support saving text transcript as PDF, HTML, TXT, WORD etc. 5 Support various languages such as English,
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    Grok Voice Think Fast 2.0
    Grok Voice Think Fast 2.0 is xAI’s flagship voice model for building real-time assistants, phone agents, and interactive voice systems that stream audio and text bidirectionally over WebSocket. Developers can configure system instructions, high or no reasoning effort, built-in or custom voices, automatic server-side voice activity detection, silence duration, idle re-engagement, playback speed, and session resumption after temporary disconnects. It accepts PCM, G.711 μ-law, G.711 A-law, and Opus audio through JSON or raw binary frames, with configurable PCM sample rates from telephone quality to 48 kHz. It supports more than 20 languages with native-quality accents, automatic language detection, natural responses in the speaker’s language, and seamless code-switching. Language hints and up to 100 key terms improve transcription of regional speech, names, products, codes, addresses, and specialized terminology, while pronunciation replacements correct spoken output.
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    Nova-3

    Nova-3

    Deepgram

    ​Deepgram's Nova-3 is an advanced speech-to-text model that sets new standards in accuracy and performance for complex, real-world scenarios. It offers real-time multilingual transcription, enabling seamless processing of conversations spanning multiple languages, a critical advancement for global customer support and emergency response services. Nova-3 also provides self-serve customization through Keyterm Prompting, allowing users to instantly adapt up to 100 domain-specific terms without the need for model retraining. This feature enhances the recognition of specialized vocabulary and technical terminology, making it highly adaptable to various industries. Additionally, Nova-3 delivers industry-leading performance with a 54.3% reduction in word error rate for streaming and 47.4% for batch processing compared to competitors. These advancements make Nova-3 a versatile solution for organizations seeking to enhance their speech recognition capabilities across diverse applications.
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    Gemini Audio
    Gemini Audio is a set of advanced real-time audio models built on Gemini's architecture, designed to enable natural, fluid voice interaction and expressive audio generation through simple language prompts. It supports conversational experiences where users can speak, listen, and interact with AI in a seamless loop, combining understanding, reasoning, and response generation in audio form. It is capable of both analyzing and generating audio, allowing applications such as speech-to-text transcription, translation, speaker identification, emotion detection, and detailed audio content analysis. They are optimized for low-latency, real-time use cases, making them suitable for live assistants, voice agents, and interactive systems that require continuous, multi-turn dialogue. Gemini Audio also integrates advanced capabilities like function calling, enabling the model to trigger external tools and incorporate real-time data into responses.
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    Cartesia Sonic-3.5
    Sonic 3.5 is Cartesia’s fastest, most natural text-to-speech model, built for expressive, real-time voice generation with sub-90ms latency and native support for 42 languages. It is designed to follow transcripts faithfully, voice confirmation codes, and heteronyms correctly without preprocessing, and stay expressive enough to carry a real conversation. It supports languages intended to deliver native-quality speech. Sonic 3.5 focuses on clean audio across every language and voice, with no artifacts to edit out, making it practical for production voice experiences where quality, speed, and consistency matter. Its expressive conversational delivery provides strong pacing and real emotional range, tuned for support and agent transcripts. Alphanumerics such as order numbers, phone numbers, IDs, and emails are spoken naturally in every language, while context-aware English pronunciation helps words like read, bass, and bow land correctly from the surrounding text.
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    Blabby

    Blabby

    Blabby

    BlabbyAI is a Chrome extension that transforms your spoken words into polished, formatted text directly inside any web text field. Once installed, it adds a discreet microphone icon to every input box (in Gmail, Docs, ChatGPT, LinkedIn, Outlook, and thousands more). Tap the icon, speak naturally, and your speech is transcribed with automatic punctuation, capitalization, and grammar correction. It supports more than 90 languages and allows users to create custom modes that tailor how their speech is converted, e.g., for emails, casual chat, or formal documents. BlabbyAI emphasizes privacy by processing voice securely without storing it after transcription. Its seamless integration across sites means you can use voice typing everywhere you type online, enabling faster writing and reducing friction from having to switch between typing and speaking.
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    MAI-Voice-2

    MAI-Voice-2

    Microsoft AI

    MAI-Voice-2 is Microsoft AI’s most expressive and natural-sounding text-to-speech model to date, built for production voice experiences where fidelity, language coverage, speaker consistency, and emotional range directly shape the user experience. It is designed for assistants, customer support, audiobooks, accessibility experiences, games, podcasts, courses, simulations, and creator workflows where voice quality must sound natural, fluid, and trustworthy. It expands from English-only support to 15 languages while maintaining naturalness and expressiveness, with support for English, Italian, French, German, Hindi, Spanish, Portuguese, Korean, Chinese, Turkish, Russian, Thai, Dutch, Romanian, and Hungarian. MAI-Voice-2 offers granular emotion control through tags such as sad, whispered, and excited, along with role-based expressive speech for experiences like motivational trainers, sports commentators, or character voices.
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    Echo Speech-to-Text

    Echo Speech-to-Text

    Echo Speech-to-Text

    Voice typing. Dictate into any website. Real-time voice transcription. Echo - Speech-to-Text is a state-of-the-art voice typing tool that works on most websites. Experience the most accurate speech recognition accuracy available. Key Features: - ✨ Automatic Punctuation: Enjoy automatic punctuation for polished, professional text. - 🗣️ Voice Type Directly into Textbox: No weird overlay or copy-pasting. - 🌍 Multi-language Support: Supports 50+ languages, including English, Spanish, German, French, etc. - 🛠️ Custom Vocabularies: Add specialized vocabulary or uncommon nouns to boost transcription accuracy. - ⌨️ Keyboard Shortcut: Start and pause voice recognition quickly with a simple keyboard shortcut. 🔒 Trusted and Secure Your privacy is our priority – we do not collect or share your data. We do NOT store any dictation text in our database. 🛡️ HIPAA Compliance We are HIPAA compliant in practice. Audio recordings are never stored. Transcription texts are
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    UntitledPen

    UntitledPen

    UntitledPen

    UntitledPen is an AI-powered platform that enables users to write, refine, and instantly transform text into realistic, human-like voice‑overs using advanced GPT-based audio generation. It features a notetaking-style smart editor and smart writing assistant to generate scripts, refine text, or polish content in any language. Users can convert text to speech or speech to text, choose from a range of voices, and customize tone, accent, and personality. Quick commands streamline writing and audio creation, while built‑in voice editing tools allow lightweight adjustments. With support for natural voice output suitable for podcasts, videos, presentations, and more, the platform includes audio download and upload options, along with smart transcription for turning speech into polished text. UntitledPen is currently in open beta and invites users to try its capabilities for free.
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    VoicePen

    VoicePen

    VoicePen

    Upload your audio or video file and VoicePen will generate a blog post + transcription using AI. The transcription + SRT file are generated with the best speech-to-text model on the market. Voicepen extracts key topics from your audio and crafts an engaging blog post. You can convert any language audio file into an English blog post. Just upload your file.
    Starting Price: $4.99 per conversion
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    Voxtral TTS

    Voxtral TTS

    Mistral AI

    Voxtral TTS is a state-of-the-art, multilingual text-to-speech model designed to generate highly realistic and emotionally expressive speech from text, combining strong contextual understanding with advanced speaker modeling to produce natural, human-like audio output. Built as a lightweight model with around 4 billion parameters, it delivers efficient performance while maintaining high quality, enabling scalable deployment for enterprise voice applications. It supports nine major languages and diverse dialects, and can adapt to new voices using only a short reference audio sample, capturing not just tone but also rhythm, pauses, intonation, and emotional nuance. Its zero-shot voice cloning capabilities allow it to replicate a speaker’s style without additional training, and it can even perform cross-lingual voice adaptation, generating speech in one language while preserving the accent of another.
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    Dictation - Voice to Text

    Dictation - Voice to Text

    Christian Neubauer

    ​Dictation - Voice to Text is an application that enables users to dictate, record, and translate text instead of typing, facilitating text generation in a 'dictation' setup with one speaker in front of the microphone. It supports more than 40 languages for dictation and over 40 languages for translation, allowing users to switch between different language projects with a single click. It offers AI-based transcription capabilities, allowing users to transcribe audio recordings, videos, voice memos, URLs, and YouTube content using OpenAI's speech recognition technology. Both audio recordings and text files can be accessed via the Apple 'Files' app and shared along with the text. With iCloud synchronization enabled, text is automatically synchronized across all devices running Dictation, including iPhone, iPad, macOS, and Apple Watch. It also supports the system font size setting and provides configurable button sizes for visually impaired users.
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    Cartesia Sonic-3
    Cartesia Sonic-3 is a real-time, streaming text-to-speech (TTS) model designed to generate ultra-realistic, expressive voice output with extremely low latency, enabling AI systems to speak as fluidly as humans in live interactions. Built on advanced state space model architecture, Sonic delivers high-quality speech while achieving near-instant response times, with audio generation beginning in as little as 40–100 milliseconds, making conversations feel seamless rather than delayed. It is optimized for conversational AI use cases, acting as the “voice layer” for AI agents by converting text into natural-sounding speech that includes emotional nuance such as excitement, empathy, or even laughter. It supports more than 40 languages with native-level voices and accent localization, allowing developers to build globally accessible applications with consistent quality across regions.
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    GPTScribe

    GPTScribe

    GPTScribe

    GPTScribe is an audio and video transcription tool built to convert speech into accurate, readable text in seconds. Users can paste a link or upload an audio or video file, and GPTScribe immediately processes the content into a transcript that can be searched, edited, scrolled, or downloaded directly in the browser. It is built on a multilingual speech model fine-tuned on noisy, real-world recordings, helping it stay accurate with overlapping voices, soft accents, background music, phone-interview hiss, coffee-shop hum, and other imperfect audio conditions. Punctuation, casing, and paragraph breaks are added automatically so the transcript reads like something a human would type instead of a wall of words. GPTScribe supports more than 100 spoken languages with automatic detection, including multilingual recordings where speakers switch languages mid-conversation.