Alternatives to Muse Voice Transcribe

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

  • 1
    Muse Spark 1.1
    Muse Spark 1.1 is a multimodal reasoning model from Meta Superintelligence Labs built for agentic tasks, coding, computer use, tool use, and multimodal understanding. The model improves on the original Muse Spark with stronger performance in planning, orchestration, long-context work, coding workflows, and external app interactions. Muse Spark 1.1 can manage a 1 million token context window, remember earlier actions, retrieve important information, compact context, and delegate tasks across parallel subagents. It is designed to operate across tools, MCP servers, custom skills, browsers, native apps, scripts, images, video, PDFs, and audio-based workflows. Developers can access Muse Spark 1.1 through the new Meta Model API public preview, while users can try it in Thinking mode in the Meta AI app and on meta.ai.
    Starting Price: $1.25 per 1M tokens (input)
  • 2
    Muse Spark 1.2
    Muse Spark 1.2 is Meta’s coding-focused model update designed to power Muse Code and improve software engineering workflows. The model is built for code generation, complex debugging, codebase understanding, long-horizon development tasks, and end-to-end developer workflows. Muse Spark 1.2 was co-trained with Muse Code to improve performance inside the terminal coding agent environment. It supports planning, goal conditioning, context compaction, subagent coordination, and iterative coding workflows across large repositories. The model was trained with expanded coding compute, diverse development environments, self-improvement loops, and long-running engineering tasks. Built for AI developers and software teams, Muse Spark 1.2 helps agents plan, write, validate, debug, and optimize code with greater autonomy.
    Starting Price: $1.25 per 1M tokens (input)
  • 3
    Muse Video
    Muse Video is Meta’s upcoming video generation model from Meta Superintelligence Labs, previewed alongside the launch of Muse Image. The model is built on the same pretraining foundation as Muse Image and is designed to generate high-fidelity videos with native audio support. Muse Video focuses on prompt adherence, visual realism, temporal consistency, and the ability to create short scenes with clear motion, continuity, and audio context. It can generate a wide range of video styles, including cinematic footage, UGC-style ads, animal scenes, product commercials, handheld point-of-view clips, and realistic moments with sound effects, voices, and music. Meta is continuing to improve areas such as audio-video synchronization and physically accurate fast motion before broader release. Coming soon to creators and Meta AI, Muse Video is positioned as a powerful tool for generating dynamic media across Meta’s creative ecosystem.
  • 4
    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.
  • 5
    MAI-Transcribe-2

    MAI-Transcribe-2

    Microsoft AI

    MAI-Transcribe-2 is Microsoft AI’s most capable transcription model yet, designed to deliver fast, accurate speech recognition across a broad range of real-world audio. It supports speaker diarization to distinguish speakers and attribute words to the right person, along with word-level timestamps for precise alignment, search, navigation, and editing. Keyword biasing helps recognize domain-specific terminology, abbreviations, names, and other terms that can be difficult to distinguish from context alone. Developers can choose between configurable transcription styles: a verbatim setting that preserves filler words and false starts for compliance and analysis, or a clean setting that removes fillers for more readable captions, notes, and published transcripts. The model supports code-switching for conversations that naturally move between languages, including blended language pairs such as Hinglish and Spanglish, and can automatically identify the language being spoken.
  • 6
    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
  • 7
    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
  • 8
    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
  • 9
    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.
  • 10
    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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    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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    Gladia

    Gladia

    Gladia

    Gladia is a speech-to-text platform built for production, turning raw audio into structured outputs that power real workflows like meeting summaries, CRM enrichment, contact center QA, and real-time voice assistants. With support for 99+ languages and the ability to handle messy real-world audio—overlapping speakers, accents, code-switching, domain-specific terminology—Gladia is designed for the complexity of actual conversations, not clean studio recordings.
    Starting Price: 10 hours free
  • 13
    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.
  • 14
    Azure Speech Translation
    Translate audio from more than 30 languages and customize your translations for your organization’s specific terms, all in your preferred programming language. Benefit from fast, reliable speech translation powered by neural machine translation technology. Generate speech-to-speech and speech-to-text translations with a single API call. Speech Translation captures the context of full sentences to provide accurate, fluent translations and improve communication between speakers of different languages. Customize speech recognition and translation for terminology specific to your business or industry. Train and deploy a custom translation system, without requiring machine learning expertise. Speech Translation can remove verbal fillers ("um," "uh," and coughs) and repeated words, add proper punctuation and capitalization, and exclude profanities for more readable translations. Deliver readable translations with an engine trained to normalize speech output.
    Starting Price: $0.36 per hour
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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.
    Starting Price: $7.50 per month
  • 16
    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.
  • 17
    RiverScript

    RiverScript

    RiverScript

    Transcribe everything you can hear on your computer Capture and turn into text everything you can hear on your computer – meetings, podcasts, any videos with Live Recording Transcription from RiverScript. Your sound – your rules. A multi-model AI architecture combining leading speech recognition models from ElevenLabs, OpenAI and Deepgram. Interactive editor, timecodes, speaker diarization. Lightning-fast desktop client for Windows and macOS, built on Rust. Supports audio and video files up to 50 GB and 8 hours long. ● works with audio and video files up to 50 GB, including batch uploads ● has a built-in editor and an interactive media player ● translates transcripts into other languages with AI ● generates subtitles with clickable timestamps ● performs speaker diarization ● creates AI-powered summaries ● lets you ask AI anything about your transcript RiverScript – transcribe everything!
  • 18
    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
  • 19
    Subanana

    Subanana

    Datax Limited

    Subanana is an AI speech-to-text web app that turns audio and video into subtitles, transcripts, and meeting summaries in 80+ languages, with standout accuracy on Asian and mixed-language speech (Cantonese, Mandarin, Japanese, Korean, and code-switching) that English-first tools handle poorly. Subtitles: import a file or a YouTube/Instagram/Facebook link, edit with a glossary and AI auto-correct, and export SRT, VTT, TXT, DOCX, bilingual subtitles, or burned-in video. Transcripts: speaker labels, filler-word removal, automatic punctuation and paragraphs. Meeting summaries: templates, decisions and action items, plus a Google Meet and Microsoft Teams recording bot that processes the meeting after it ends. Live captions: real-time captioning with translation for events.
  • 20
    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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    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
  • 22
    FastScribe

    FastScribe

    FastScribe

    AI transcription tool that converts audio and video to text with timestamps and automatic speaker labels. Identifies who is speaking and separates the transcript into labelled turns, which you can rename. Supports MP3, M4A, WAV, AAC, FLAC, OGG, Opus, WMA, AMR, MP4, MOV, WEBM, AVI, MKV and more. Exports TXT, SRT, VTT and DOCX subtitles with speaker names included. Free tier with no signup required for the first file, speaker labels included. Speech recognition and speaker diarization both run on private self-hosted GPU hardware, and audio is deleted immediately after transcription. Supports Spanish, French, German, Portuguese, Italian, Japanese, Hindi, Korean and more.
    Starting Price: $12/user/month
  • 23
    Ecango

    Ecango

    Ecango

    Ecango is an AI-powered audio and video transcription tool that converts spoken content into accurate, searchable text in seconds. Users can upload or drag and drop audio or video files, let Ecango generate the transcript, then edit it directly in the browser and export it in popular formats including DOCX, ODT, PDF, SRT, and TXT. It supports transcription, subtitles, and translation across more than 90 languages, dialects, and accents, using advanced speech recognition to deliver up to 99.8% accuracy. Speaker identification and diarization detect different people speaking within the same recording and organize their dialogue into an easy-to-read transcript. Ecango supports popular audio and video formats and automatically handles video files without requiring users to separate the audio first. Its AI can also filter background noise to improve transcription and translation results when recordings are less than ideal.
    Starting Price: $99 per month
  • 24
    Speakwise

    Speakwise

    Speakwise

    Speakwise is an AI voice recorder, voice-to-text transcription app, and note taker for iPhone built for in-person meetings and conversations. With one tap, users can record a meeting directly through the iPhone microphone without a laptop, Zoom, Teams, calendar invite, or bot joining the call. Speakwise converts the recording into a searchable transcript, automatically labels speakers with diarization, and generates a clean AI summary with key points, decisions, and action items. It supports more than 100 languages with automatic detection and dialect recognition, including regional speech that other note takers may struggle with. Recording works offline, so conversations can be captured on planes, job sites, secure rooms, and other places without Wi-Fi, then synced and processed when a connection returns. AirPods controls let users start, pause, and stop recording hands-free, while the iPhone Action Button can launch recording instantly on supported devices.
  • 25
    Baidu AI Cloud Speech-to-Text
    Baidu’s speech technology provides developers with such industry-leading capabilities as speech-to-text,text-to-speech, and speech wake-up. Combining with the NLP technology, it is applicable for several scenarios, including speech input, speech search, video subtitle, audio content analysis, calling center, book broadcasting, news broadcasting, and order broadcasting. It can convert a speech with a duration of fewer than 60 seconds to characters. It is applicable for mobile speech input, intelligent speech interaction, speech commands, and speech search. It can convert the audio stream into characters and return each sentence's start and end times. It is applicable for such scenarios as long-sentence speech input, audio and video subtitles, and meeting records. It can convert the audio files uploaded in batches into characters and return the recognition results within 12 hours. It is applicable for such scenarios as record quality check, and audio content analysis.
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    Higgs Audio / Avatar
    Higgs Audio / Avatar is a family of foundation audio and avatar models designed to generate natural speech, understand tone, emotion, and intent, and give voice interactions a visual presence. The models support text-to-speech, speech-to-text, avatar generation, and automatic voice casting that selects an appropriate voice based on context, sentiment, and content. Built for real-world production, Higgs combines expressive generation, robust speech understanding, and flexible deployment for workloads where quality, latency, and reliability matter. High-accuracy multilingual speech recognition supports major languages, while voice cloning reproduces a speaker’s tone from short reference samples to maintain consistent brand voices across interactions. Sentiment detection reads emotional signals in speech to enable smarter routing, stronger analytics, and more context-aware agent behavior.
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    Voicetapp

    Voicetapp

    Voicetapp

    convert speech to text quickly and accurately with over +170 languages & dialects. Speaker Identification Feature allows you to identify up to 5 speakers in the audio. Our enhanced live transcribe feature allow you to use 12 languages to transcribe audio in real time. Voicetapp have a super clean & easy to use dashboard, to make users very confortable while using it. Thanks to deep learning tecknology supported by AI, we can guarantee up to 100% accuracy rates. Our enhanced ASR engine, powered by its detection and interpretation capabilities, can automatically identify punctuation. With our speech-to-text technology, we are changing the way people do their businesses.
    Starting Price: $9 per 60 minutes
  • 28
    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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    VideoToWords.ai

    VideoToWords.ai

    VideoToWords.ai

    VideoToWords.ai is an AI‑powered transcription tool that converts audio and video into text with 99.9% accuracy, supporting more than 98 languages and speaker recognition. Users can upload files up to ten hours in length, MP3, WAV, MP4, AVI, MPEG, M4A, and more, directly in the browser, and transcription begins automatically. It provides ultra‑fast, GPU‑accelerated processing, AI‑generated summaries for quick insights, and an intuitive online editor for reviewing and optimizing transcripts. Completed text can be exported in TXT, DOCX, PDF, SRT, or VTT formats for easy sharing, subtitle creation, or further editing. Built on industry‑leading speech and video recognition models, VideoToWords.ai ensures ironclad data security and privacy, handling meeting recordings, lectures, interviews, podcasts, and marketing content seamlessly. With extended file support, customizable export options, and global language coverage.
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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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    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.
    Starting Price: $4,000 per year
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    FluidVoice
    FluidVoice is a free, open source macOS dictation app that pairs local speech recognition with Fluid-1, an on-device AI model for polishing dictation. One hotkey lets users speak into any text field across email, documents, chat, terminals, code editors, and other apps, with text appearing in near real time. Local speech models keep dictation on-device and work offline, while optional AI post-processing can use Fluid Intelligence, OpenAI, Groq, or custom providers. Fluid-1 cleans up rough dictation, fixes formatting, capitalization, dates, names, and numbers, and adapts tone to the active app without changing the speaker’s meaning. Users can create custom prompts for different apps, while Write Mode, Command Mode, and Direct Dictation make it easy to switch contexts. FluidVoice supports more than 40 languages across models including Nemotron Speech 3.5, Parakeet Flash, Parakeet TDT v2 and v3, Cohere Transcribe, Apple Speech, and Whisper.
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    Converse Smartly
    Converse Smartly® is a powerful speech to text software which converts audio to text. It enables organizations and individuals to work smarter, faster and with greater accuracy. The application can be used to analyze dialogue or speech from team meetings, interviews, conferences and seminars. We strive to provide the preeminent online speech recognition tool by engaging cutting-edge speech-recognition technology for the most accurate results technology can achieve today, together with incorporating built-in tools to increase users' efficiency, productivity and comfort. Render the most advanced deep-learning neural network algorithms to the audio subject for speech recognition with unparalleled accuracy. Converse Smartly(s) Speech-to-Text accuracy improves over time as the continuous machine learning powered by enhanced algorithms improves the internal speech recognition technology used by multiple products.
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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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    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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    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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    Temi

    Temi

    Temi

    Upload any audio or video file. We accept all file types. Review your transcript with timestamps and speakers. Save & export your transcript as MS Word, PDF, SRT, VTT and more. Transcript quality depends on audio quality. Record clear audio to get accurate transcripts. Temi's free transcription editor lets you edit your transcripts online in minutes. Built by our machine learning and speech recognition experts. Quickly clean-up the provided transcript. Adjust the playback speed and skip around easily. Temi knows the timing of every word. Add any timestamps. We mark the change of every speaker and label them. Download your transcript into text (MS Word, PDF) or closed caption files (SRT, VTT).
    Starting Price: $0.25 per audio minute
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    Meta Model API
    Meta Model API is a new developer API for building with Muse Spark 1.1, Meta’s multimodal reasoning model built for agentic tasks, coding, tool use, computer use, and multimodal understanding. Now in public preview, it gives developers a way to access Muse Spark 1.1 through an OpenAI-compatible package, making it easier to point existing clients at the API, keep the same code structure, and set the model to muse-spark-1.1. Muse Spark 1.1 is designed for personal agentic tasks that require planning and orchestration across external apps and services, with the ability to generalize to new native tools, MCP servers, and custom skills. As a main agent, it can gather context, make a plan, and delegate execution across parallel subagents; as a subagent, it follows its role, understands available tools, and knows when to escalate back. The model can actively manage a 1 million-token context window, remember actions, retrieve information from much earlier work, and compact context.
    Starting Price: $1.25 per 1M tokens
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    Muse Image
    Muse Image is Meta’s image generation model from Meta Superintelligence Labs, built into Meta AI for creating, editing, and sharing high-quality visuals. The model can turn simple conversational prompts into detailed images, blend multiple photos together, remove unwanted objects, generate legible text inside visuals, and create styled outputs such as portraits, posters, stickers, room redesigns, infographics, and fantasy scenes. Muse Image uses advanced reasoning through Muse Spark to plan layouts, understand context, look up real-time web information, and combine visual references more intelligently. Users can start with suggested presets, mention Instagram accounts to personalize creations, and sketch or annotate edits directly on top of an image. The model powers creative experiences across Meta AI, Instagram Stories, WhatsApp chats, and soon Facebook, Messenger, and advertiser tools through Meta Advantage+ creative.
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    Gemini 3.5 Live Translate
    Gemini 3.5 Live Translate is Google’s latest audio model for live speech-to-speech translation, delivering near real-time translation in more than 70 languages. The model automatically detects multilingual input and generates smooth, natural-sounding translated speech that preserves the speaker’s intonation, pacing, and pitch. Unlike turn-by-turn translation systems that wait for someone to finish speaking before responding, Gemini 3.5 Live Translate processes speech as it streams and generates translated audio continuously, balancing the need for context with the need to stay in sync. It stays only a few seconds behind the speaker throughout a session, helping conversations feel more fluid and natural, without awkward pauses. It is built for multilingual calls, meetings, lessons, broadcasts, live interpretation, dubbing, simultaneous translation, and voice translation applications.
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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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    Higgs Realtime
    Higgs Realtime is a production-quality, real-time speech-to-speech model and API built for natural, continuous conversation. It is an end-to-end, instruction-tuned, audio-native model that can understand audio, text, or both and generate high-quality responses, while also functioning as a text LLM when given text alone. Designed for live voice agents, it follows conversations, handles interruptions, adapts when requests change mid-sentence, and carries multi-step workflows through to completion. The model is trained specifically for voice-agent reflexes such as natural turn-taking, conversational cadence, tone adaptation, spoken tool preambles, multi-turn state tracking, and robust instruction following through changing requests. Semantic turn detection helps distinguish a completed turn from a pause, while multilingual and code-switched understanding supports more than 100 languages without per-language setup.
    Starting Price: $0.0023 per minute
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    GoVivace

    GoVivace

    GoVivace

    Our automatic speech recognition engine supports several English accents and can be localized to any language. Also, the ASR engine supports standard telephony as well as web and mobile applications. Being capable of actioning voice commands given to electronic devices such as computers, tablets, smartphones or telephones with the aid of a microphone, the GoVivace’s Automatic Speech Recognition Engine finds use in diverse applications. This automatic speech recognition engine compares the spoken input with a number of pre-specified possibilities and convert speech to text. The entire set of pre-specified possibilities constitute the application’s grammar, which powers the interface between the dialogue-speaker and the back-end processing. GoVivace’s patented Automatic Speech Recognition solution needs only very simple grammar for its processing. It can also support very large grammars for complex tasks.
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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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    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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    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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    MacWhisper

    MacWhisper

    MacWhisper

    MacWhisper is an all-in-one Mac transcription app for transcribing files, meetings, lectures, podcasts, videos, subtitles, voice memos, and private recordings. The app lets users drag and drop audio or video files, record online meetings, capture app audio, and use real-time dictation in any app. MacWhisper supports Zoom, Teams, Webex, Skype, Discord, and other meeting platforms without requiring bots to join calls. Its local AI models help users transcribe sensitive files offline so data does not have to leave the Mac. The platform includes speaker recognition, filler-word removal, translation, transcript search, editing, batch transcription, exports, summaries, chat, and custom AI prompts. Built for professionals, students, creators, researchers, journalists, and privacy-conscious users, MacWhisper helps turn speech and media into clean, searchable, editable text.
    Starting Price: €59 one-time payment
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    Muse Glimmer
    Muse Glimmer is a 30-billion-parameter open-weights model from Meta Superintelligence Labs, optimized for always-on local agent workflows. Small enough to run on a Mac or PC with a single consumer GPU, it is designed for local agents, function calling, coding, and LLM-as-a-judge evaluation without depending on cloud infrastructure or network access. The model combines long-horizon execution, precise tool calling, multimodal understanding, long-context memory, and instruction following. It can complete end-to-end agentic tasks, sustain multi-step reasoning across extended workflows, recover from failed or unexpected tool calls, and accept interleaved text and images through a dedicated perception encoder for interpreting screenshots, charts, and documents. Muse Glimmer works with OpenClaw and other agentic orchestration patterns, supports controllable reasoning effort, and is trained on data from more than 100 languages.
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    EKHOS AI

    EKHOS AI

    EKHOS AI

    EKHOS AI is a secure offline transcription software developed for professionals who work with sensitive audio data. It performs accurate speech-to-text conversion without relying on cloud services, ensuring that all files remain local and private. Designed with legal, medical, academic, and research use cases in mind, EKHOS AI supports common audio formats and offers features such as timestamped transcriptions, multi-speaker diarization, segment tagging, and export to multiple text formats. An intuitive editor is included to review and refine transcripts directly within the app. The software also supports real-time audio recording and playback. EKHOS AI is built to perform reliably on a wide range of Windows systems, offering practical functionality for users who prioritize data control, security, and data privacy.
    Starting Price: $9/user/month - annual billing
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    Deepgram

    Deepgram

    Deepgram

    Deploy accurate speech recognition at scale while continuously improving model performance by labeling data and training from a single console. We deliver state-of-the-art speech recognition and understanding at scale. We do it by providing cutting-edge model training and data-labeling alongside flexible deployment options. Our platform recognizes multiple languages, accents, and words, dynamically tuning to the needs of your business with every training session. The fastest, most accurate, most reliable, most scalable speech transcription, with understanding — rebuilt just for enterprise. We’ve reinvented ASR with 100% deep learning that allows companies to continuously improve accuracy. Stop waiting for the big tech players to improve their software and forcing your developers to manually boost accuracy with keywords in every API call. Start training your speech model and reaping the benefits in weeks, not months or years.