Alternatives to Nemotron 3.5 Transcribe

An API powered by Google's AI technology allows you to accurately convert speech into text. You can accurately caption your content, provide a better user experience with products using voice commands, and gain insight from customer interactions to improve your service. Google's deep learning neural network algorithms are the most advanced in automatic speech recognition (ASR). Speech-to-Text allows for experimentation, creation, management, and customization of custom resources. You can deploy speech recognition wherever you need it, whether it's in the cloud using the API or on-premises using Speech-to-Text O-Prem. You can customize speech recognition to translate domain-specific terms or rare words. Automated conversion of spoken numbers into addresses, years and currencies. Our user interface makes it easy to experiment with your speech audio.

Rev is an Investigative Intelligence Platform built for legal, law enforcement, court reporting, and investigative workflows. The platform helps teams turn audio, video, documents, police reports, depositions, body cam footage, medical records, and case files into searchable and citable records. Rev combines AI transcription, human transcription, evidence analysis, document editing, image analysis, AI templates, clipping, and secure dictation. Users can ask direct questions across evidence files to identify contradictions, reconstruct timelines, find key moments, and support case preparation. Every AI-generated answer is tied back to the original record so teams can verify findings instead of relying on unsupported model output. Rev also helps users turn findings into memos, outlines, case summaries, motions, trial briefs, affidavits, and other legal work product. Its transcript editor allows teams to mark up testimony, create timestamped clips, and securely share evidence with trial teams. Rev emphasizes security with encryption, legal workflow controls, and a policy that uploaded data is not sold or used to train third-party LLMs. By combining transcription, evidence search, AI analysis, citations, secure collaboration, and legal drafting workflows, Rev helps investigative teams find critical facts faster.

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 🚀 Power your Speech-to-Text and Voice AI with Speechmatics today!

Cartesia Ink represents a suite of real-time streaming speech-to-text (STT) models that facilitate swift and natural dialogues within voice AI applications by serving as the essential “voice input” layer that transforms spoken words into precise text without delay. Its premier model, Ink-Whisper, is meticulously crafted for conversational settings, providing transcription with an impressively low latency of just 66 milliseconds, which fosters seamless, human-like communication free from noticeable interruptions. In contrast to conventional transcription methods designed for batch processing, Ink is tailored for live interactions, adeptly managing fragmented and varied audio through an innovative dynamic chunking approach that minimizes errors and enhances responsiveness, particularly during pauses, interruptions, or brisk exchanges. Consequently, this advanced technology ensures that users experience a smoother and more engaging interaction, reflecting the evolving demands of modern communication.

NVIDIA Nemotron 3.5 Bolt is a series of efficient AI models developed by NVIDIA for running autonomous agents and conversational applications directly on local hardware. The family consists of Bolt 74B-A9B 24GB and Bolt E4B 4GB, which address different performance and memory requirements. Bolt 74B-A9B uses a mixture-of-experts architecture with 74 billion total parameters and approximately 9 billion active parameters to balance reasoning capabilities with computational efficiency. Native 2-bit quantization enables the model to operate within a 24GB memory footprint on supported NVIDIA systems. Its agentic capabilities include web research, code generation, software debugging, information retrieval, and multistep task execution. The model can be integrated with local AI agent frameworks and development tools to complete assignments without sending application workloads to external inference services. Bolt E4B is optimized for fast conversational responses, multistep interactions, and embedded AI experiences such as virtual assistants and interactive game characters. NVIDIA supports local deployment of the Bolt family across compatible GeForce RTX, RTX PRO, DGX Spark, and Jetson hardware, depending on the model variant. Bolt 74B-A9B is available with open weights through Hugging Face and supports inference tools including llama.cpp, Ollama, LM Studio, Unsloth, and vLLM.

Grok Speech to Text is an independent audio API created to assist developers in seamlessly incorporating quick and precise transcription capabilities into various applications. Utilizing the same technology framework that drives Grok Voice, Tesla vehicles, and Starlink's customer support services, this API caters to multiple applications such as voice assistants, real-time transcription solutions, accessibility enhancements, podcasts, meeting documentation, telephony, and engaging audio experiences. Grok STT is capable of producing transcripts from extensive audio files via a REST API or transcribing speech instantly using a low-latency WebSocket API. It features word-level timestamps, speaker differentiation, support for multiple audio channels, and advanced Inverse Text Normalization, which transforms spoken language into correctly formatted structured outputs for different data types, including numbers, dates, and currencies. Grok Speech to Text has been rigorously tested across various formats, including phone calls, meetings, videos, and podcasts, demonstrating exceptional accuracy in entity recognition and various business applications. This API provides a versatile solution for developers looking to enhance their application's audio capabilities with reliable transcription features.

Gemini 3.5 Transcribe represents Google’s most advanced speech-to-text technology to date, tailored for sophisticated voice interactions and immediate transcription. Rather than merely translating speech into text, it converts raw audio into polished, precise, and well-structured text, effectively managing background noise, intricate terminology, various accents, dialects, and natural speech rhythms. Its intelligent transcription capabilities automatically account for self-corrections, eliminate filler words like “ums” and “ahs,” and present the final output in an easily readable format. This model offers continuous bidirectional streaming with sub-second response times, making it ideal for interactive voice applications, alongside the ability to process pre-recorded audio for meetings, call logs, and other recordings while ensuring speaker attribution and word-level timestamps. Additionally, its custom vocabulary feature enables the recognition of specialized terms, unique spellings, postal codes, order IDs, and other industry-specific language, enhancing its versatility for various use cases. As a result, Gemini 3.5 Transcribe stands out as a powerful tool for anyone seeking high-quality transcription services.

Voxtral Mini Transcribe 3, developed by Mistral AI, is an advanced speech-to-text model that transforms audio recordings into written transcripts via an API. Developers have the ability to submit audio files for bulk transcription, receiving text that includes timestamps and identifies different speakers. It provides support for transcription in multiple languages, incorporates speaker diarization, and allows for context biasing, enabling applications to introduce specific names or specialized terminology to enhance recognition accuracy. This model is specifically tailored for developers who are creating applications and audio workflows that require making recorded speech easily searchable, attributing dialogue to specific speakers, or synchronizing transcript segments with their corresponding audio. Teams can access the model through Mistral's hosted API, seamlessly integrating the resulting output into their software solutions, including transcription workflows and voice-responsive services. Unlike typical recording or editing applications, this offering is focused solely on the model and API functionalities. Furthermore, its capabilities make it a valuable asset for various industries aiming to streamline their audio data management processes.

MAI-Transcribe-2 represents the pinnacle of Microsoft AI's transcription capabilities, engineered to provide rapid and precise speech recognition across various real-world audio scenarios. This model includes features like speaker diarization, enabling it to differentiate between speakers and correctly attribute dialogue, as well as offering word-level timestamps for enhanced alignment, searching, navigation, and editing purposes. Additionally, it utilizes keyword biasing to improve the recognition of specialized terms, abbreviations, and names that may otherwise be challenging to identify from their contextual usage. Developers are afforded the flexibility to select from different transcription styles: a verbatim option that retains filler words and false starts for thorough analysis and compliance, or a clean option that eliminates such elements for clearer captions and more polished published transcripts. Furthermore, the model is adept at handling code-switching, seamlessly transitioning between languages during conversations, even accommodating mixed language combinations like Hinglish and Spanglish, while automatically identifying the language being spoken. This makes MAI-Transcribe-2 an invaluable tool for diverse linguistic environments and applications.

Voxtral Small Transcribe 3 is a versatile multilingual speech-to-text model tailored for converting recorded audio into text. Offered via an audio transcription API by Mistral, it incorporates features like timestamp documentation, speaker diarization, and context biasing, enabling developers to input key phrases that enhance the recognition of specific terms and names relevant to particular domains. This model is capable of automatically identifying supported languages or can utilize a specified language code, accommodating over 50 languages in total. It is particularly suitable for application developers focused on creating workflows for transcribing recorded audio, such as meetings, interviews, call transcriptions, customer service systems, and voice-agent processes. Additionally, Mistral outlines options for self-deployment of the Voxtral 3 series, which includes hosting on private clouds or on-premises solutions, making it adaptable to various organizational needs. Furthermore, the flexibility of the model ensures that it remains a valuable tool for enhancing communication efficiency across different sectors.

Voxtral Mini Transcribe Realtime 3 is an advanced speech-to-text solution offered via Mistral’s API, designed for applications requiring immediate transcription capabilities. It utilizes WebSocket technology to stream audio and delivers transcription events in real-time as individuals speak, with adjustable streaming latencies. This model is ideal for developers looking to incorporate live captions, enhance voice assistant functionalities, or streamline voice-agent processes with accurate transcription. Notably, it includes features such as turn detection, speaker diarization, and context biasing tailored for specialized vocabulary, and it accommodates transcription in 45 different languages and dialects. Users can connect seamlessly to the real-time endpoint, allowing them to receive transcription events as audio is processed, eliminating the need to wait for a complete audio recording to finish. As a member of Mistral’s Voxtral audio model family, Voxtral Mini Transcribe Realtime 3 stands out for its versatility and efficiency in a variety of applications. This capability makes it an essential tool for developers aiming to enhance user interaction through immediate and accurate voice transcription.

Easily and efficiently develop voice-enabled applications with the Speech SDK, which allows for precise speech-to-text transcription, the generation of realistic text-to-speech voices, and the translation of spoken audio while also incorporating speaker recognition features. By utilizing Speech Studio, you can design customized models that suit your specific application needs, benefiting from advanced speech recognition, lifelike voice synthesis, and award-winning capabilities in speaker identification. Your data remains private, as your speech input is not recorded during processing, and you can create unique voices, expand your base vocabulary with specific terms, or develop entirely new models. The Speech SDK can be deployed in various environments, whether in the cloud or through edge computing in containers, enabling rapid and accurate audio transcription across more than 92 languages and their respective variants. Furthermore, it provides valuable customer insights through call center transcriptions, enhances user experiences with voice-driven assistants, and captures critical conversations during meetings. With options for text-to-speech, you can build applications and services that engage users conversationally, selecting from an extensive array of over 215 voices in 60 different languages, making your projects more dynamic and interactive. This flexibility not only enriches the user experience but also broadens the scope of what can be achieved with voice technology today.

Whisper is a powerful speech-to-text model created by OpenAI to deliver accurate and reliable audio transcription. It is trained on a large dataset of 680,000 hours of multilingual audio, making it highly robust across different languages and environments. The model performs multiple tasks, including transcription, translation, and language detection within a single system. Whisper uses a Transformer-based encoder-decoder architecture to process audio converted into log-Mel spectrograms. It can generate phrase-level timestamps and handle noisy or complex audio inputs effectively. Unlike many specialized models, Whisper is designed for strong zero-shot performance across diverse datasets. It supports multilingual transcription and can translate speech from various languages into English. The model is open-sourced, allowing developers and researchers to build and customize applications بسهولة. Its flexibility makes it suitable for use cases like voice assistants, transcription services, and accessibility tools. Overall, Whisper provides a scalable and versatile foundation for speech processing applications.

Voxtral models represent cutting-edge open-source systems designed for speech understanding, available in two sizes: a larger 24 B variant aimed at production-scale use and a smaller 3 B variant suitable for local and edge applications, both of which are provided under the Apache 2.0 license. These models excel in delivering precise transcription while featuring inherent semantic comprehension, accommodating long-form contexts of up to 32 K tokens and incorporating built-in question-and-answer capabilities along with structured summarization. They automatically detect languages across a range of major tongues and enable direct function-calling to activate backend workflows through voice commands. Retaining the textual strengths of their Mistral Small 3.1 architecture, Voxtral can process audio inputs of up to 30 minutes for transcription tasks and up to 40 minutes for comprehension, consistently surpassing both open-source and proprietary competitors in benchmarks like LibriSpeech, Mozilla Common Voice, and FLEURS. Users can access Voxtral through downloads on Hugging Face, API endpoints, or by utilizing private on-premises deployments, and the model also provides options for domain-specific fine-tuning along with advanced features tailored for enterprise needs, thus enhancing its applicability across various sectors.

Rev AI is a developer-first speech-to-text API that delivers accurate transcription for prerecorded files and real-time audio streams. The platform is built for high accuracy, fast performance, and global scale across more than 57 languages. Rev AI’s speech recognition models are trained using a carefully selected subset of more than 7 million hours of human-verified speech data. The platform is designed to provide proper grammar, punctuation, formatting, and low word error rates across a wide range of use cases. Rev AI also emphasizes fairness and accuracy across ethnic backgrounds, nationalities, genders, and accents. Developers can integrate quickly using APIs, SDKs, documentation, and expert support, with cloud and on-prem deployment options. AI Insights extend transcription with language identification, sentiment analysis, topic extraction, summarization, and translation. Precision timestamps and forced alignment provide word-level timing for media, accessibility, search, and content indexing. By combining speech-to-text, real-time transcription, global language coverage, AI insights, timestamps, and enterprise-grade security, Rev AI helps teams unlock more value from voice data.

LFM3 represents Liquid AI's suite of streamlined, open-weight foundational models designed for a variety of applications, including text, vision, audio, and multimodal tasks. This lineup features variants for text generation, vision-language processing, audio, omni tasks, and decision-making models. Developers have the option to download the model weights, customize the supported models via fine-tuning, and deploy them on local edge hardware, or alternatively, utilize Liquid AI’s hosted API for access. Potential applications encompass conversational agents, tool invocation, speech transcription and synthesis, understanding of images and videos, document extraction, classification, routing, and robotics functionalities. LFM3 aims to cater to developers, researchers, and organizations focused on creating tailored AI solutions that require adaptable deployment options, including the ability for local inference. Furthermore, the open-weight models operate under Liquid AI’s LFM Open License, where the eligibility for commercial use and its associated terms are contingent upon the annual revenue of the organization. This flexibility allows users to tailor their AI implementations precisely to their needs, enhancing the overall utility of the models.

Transcribe significantly reduces the time spent on transcription each month for journalists, lawyers, podcasters, students, and professional transcriptionists globally, potentially saving thousands of hours. Boost your efficiency and reclaim valuable time by transforming a wide variety of audio content, including interviews, lectures, speeches, and podcasts, into written text. Simply put on your headphones, play your audio at a slower pace, and articulate what you hear—it's really that straightforward. Our dictation technology allows for real-time speech-to-text conversion, offering a speedier alternative to traditional typing methods. We cater to a diverse range of languages, including English, Spanish, French, Hindi, and nearly all other languages from Europe and Asia, making transcription accessible for a global audience. This versatility ensures that users from different linguistic backgrounds can benefit from our service seamlessly.

Dragon Legal is a specialized speech recognition tool designed specifically for those in the legal field, boasting a legal-centric language model crafted from an extensive database of over 400 million words derived from legal texts. This advanced software allows lawyers and legal experts to dictate documents such as contracts, briefs, and citations with impressive accuracy levels reaching up to 99%, and at a speed that is three times quicker than traditional typing methods. Users can also create personalized voice commands to streamline repetitive tasks and benefit from the ability to transcribe previously recorded audio, significantly boosting overall workflow efficiency. Dragon Legal v16 is optimized for Windows 11 and remains compatible with Windows 10, while also offering features that enhance accessibility, including the ability to playback dictated text and utilize advanced macro commands for professionals who may face physical or cognitive challenges. Furthermore, it seamlessly integrates with Dragon Anywhere Mobile, a cloud-based dictation service for both iOS and Android devices, allowing legal practitioners to maintain their productivity even while on the move. This combination of features ensures that legal professionals can work more effectively in their demanding environments.

Efficiently and precisely convert audio into text across over 85 languages and their variations. Enhance transcription accuracy by customizing models to better suit specific industry jargon. Unlock the full potential of spoken audio by allowing for search capabilities or analytics on the transcribed text, or enabling actions through your chosen programming language. Achieve high-quality audio-to-text transcriptions through advanced speech recognition technology. Expand your base vocabulary by incorporating particular terms or create your own bespoke speech-to-text models. Operate Speech to Text in various environments, whether in the cloud or locally through containers. Leverage the powerful technology that supports speech recognition in Microsoft products. Transform audio input from diverse sources, including microphones, audio files, and blob storage. Utilize speaker diarisation techniques to identify who spoke and when. Obtain well-structured transcripts complete with automatic punctuation and formatting. Customize your speech models for a better understanding of terminology specific to your organization or industry, ensuring a higher level of accuracy in your transcriptions. This versatility makes it easier to adapt the technology to your specific needs and applications.

Phonon serves as an open-weight text-to-speech (TTS) model and inference runtime designed for developers seeking to incorporate speech generation into applications that require local and offline audio capabilities. This innovative system transforms text into spoken audio and has the ability to stream speech in real-time as it is produced. The runtime offers interfaces for Rust and Python, a Swift package for iOS and macOS, JavaScript support for web browsers, and libraries for Android; it can also function as a self-hosted server compatible with OpenAI's API. Depending on the variant of the model used, it accommodates languages such as English, French, German, Spanish, and Portuguese. Phonon is tailored for developers aiming to integrate speech functionalities into mobile, desktop, web, embedded, and server applications, particularly in scenarios requiring offline access or enhanced privacy. The English model, which consists of 31 million parameters, is licensed under Apache 2.0, while the language-specific and multilingual models, featuring 40 million parameters, operate under CC BY-NC 4.0. Additionally, the runtime’s code is available under either the Apache 2.0 or MIT licenses, making it accessible for various development needs. This flexibility and wide-ranging support make Phonon a versatile choice for speech synthesis across different platforms.

Convert audio and video files into written text effortlessly. Achieve high-quality transcriptions for podcasts utilizing specialized speech recognition tailored to specific industries. SpeechText.AI stands out as an advanced software solution designed for transforming spoken content into text format. Users can easily upload their audio or video files and benefit from AI transcription that accommodates various formats and languages. Choose your relevant domain and audio type from established categories to enhance the accuracy of transcribing industry-specific terminology. Upon selecting the appropriate settings, the sophisticated transcription engine employs cutting-edge deep neural network models to produce text that closely resembles human accuracy. Additionally, users can interactively edit, search, and validate their transcriptions using intuitive editing tools, with the flexibility to export the final content in multiple formats. The array of exceptional features within SpeechText.AI ensures that audio and video transcription is accomplished in mere seconds, thanks to its robust speech recognition capabilities. With its user-friendly interface and advanced technology, SpeechText.AI is poised to meet all your transcription needs.

Diktamen is an innovative cloud-based platform for digital dictation and transcription aimed at enhancing voice capture, task management, and workflow automation across various professional fields. Users can dictate audio from virtually anywhere—whether through mobile devices, desktops, or specialized equipment—and securely send that audio for transcription, speech recognition, and task allocation. The platform is tailored to meet the specific needs of industries such as legal and healthcare, seamlessly integrates with existing systems, and offers centralized management for submission oversight, status monitoring, and business intelligence reporting, all powered by AI-driven forecasting. By utilizing Diktamen, clients can significantly lower their dictation infrastructure costs, experience quicker transcription turnaround via outsourced partner networks, and benefit from real-time task routing. Additionally, the platform’s flexible SaaS deployment model requires minimal local installation and maintenance, making it user-friendly. Diktamen also boasts ISO 27001 certification and complies with GDPR regulations to ensure data security and adherence to compliance standards. This comprehensive approach not only enhances operational efficiency but also provides peace of mind regarding data protection.

Gladia is an advanced audio transcription and intelligence solution that provides a cohesive API, accommodating both asynchronous (for pre-recorded content) and real-time transcription, thereby allowing developers to translate spoken words into text across more than 100 languages. This platform boasts features such as word-level timestamps, language recognition, code-switching capabilities, speaker identification, translation, summarization, a customizable vocabulary, and entity extraction. With its real-time engine, Gladia maintains latencies below 300 milliseconds while ensuring a high level of accuracy, and it offers “partials” or intermediate transcripts to enhance responsiveness during live events. Overall, Gladia stands out as a versatile tool for developers looking to integrate comprehensive audio transcription capabilities into their applications.

Gradium offers a voice AI platform tailored for developers and businesses that are creating applications with speech capabilities and conversational interfaces. The platform's hosted APIs encompass a variety of functions, including streaming text-to-speech, speech-to-text, speech translation, and speech-to-speech, accessible via both WebSocket and REST interfaces, along with features like real-time turn detection, pronunciation adjustments, and voice design or cloning options. Users can explore the models and manage voices through a web-based studio, while comprehensive documentation and SDK guides aid in the development of applications. Potential applications span across voice agents, automation in customer service, live transcription services, multilingual voice interactions, and the generation of spoken content. Additionally, Gradium has introduced Phonon, an open-weight text-to-speech model that is intended for offline use in applications and on-device systems. The platform operates on a credit-based system for various services such as speech generation, transcription, and translation, ensuring flexibility and scalability for users. This comprehensive approach makes Gradium a versatile choice for those looking to enhance their applications with advanced voice capabilities.

Soniox creates advanced foundational speech models that facilitate real-time transcription, translation, and comprehension of spoken language, while also offering a developer platform that simplifies the integration of real-time voice intelligence into various applications. Their Speech-to-Text API enables users to transcribe spoken content in over 60 languages with impressive accuracy, designed for large-scale use. Additionally, Soniox ensures regional data residency and adheres to compliance standards such as SOC 2 Type 2, GDPR, and HIPAA, making it a reliable choice for businesses. This commitment to compliance and security enhances trust in their services, allowing companies to utilize voice technology confidently.

Dragon Professional is an advanced speech recognition tool designed to help professionals generate high-quality documents more effectively by turning spoken words into text with an impressive accuracy rate of up to 99%. Tailored for Windows 11 and also compatible with Windows 10, it caters to a wide range of industries, including finance, education, and healthcare. Users can dictate their documents three times more rapidly than they could type, and the software also supports the transcription of pre-recorded audio files. Moreover, it features customizable options, allowing users to create specific words and commands that can enhance efficiency by minimizing repetitive tasks. In addition, Dragon Professional v16 provides users with access to Dragon Anywhere Mobile, a convenient cloud-based dictation service available for iOS and Android devices, which facilitates productivity while on the move. This innovative software not only improves workflow but also empowers users to leverage technology for better document management.

Utilize our automatic speech-to-text technology to transcribe your content, or opt for manual transcription or professional services if preferred. Our online time-synchronous editor allows you to navigate seamlessly through your data and corresponding transcripts. You can download your transcripts in various file formats for added convenience. Organize your team of transcribers efficiently using tags and categories, while providing them support through our automatic voice-to-text capabilities. Integrate SpokenData into your applications via our REST API, which is designed to enhance the transcription accuracy by tailoring the voice-to-text functionality to your specific data domain, ultimately reducing labor costs. By enabling speech technologies within your applications through our API, you can confidently handle large volumes of data. We offer a customizable API that aligns with your unique requirements, and our support team is ready to assist you. Our voice-to-text solutions are specifically adapted to your data and its intended use, ensuring optimal accuracy in your transcripts. This service is ideal for web and mobile app developers, media monitoring agencies, and businesses involved in audio or video archiving, making it a valuable resource across various industries. Additionally, our commitment to precision and customization will enhance the overall efficiency of your transcription processes.

Effortlessly generate transcripts, subtitles, and voiceovers in mere minutes with state-of-the-art speech-to-text software featuring an integrated advanced text editor. This tool supports translation in English, French, Spanish, German, and over 80 other languages. Save both time and resources through Maestra’s automatic audio transcription capabilities, which convert audio files to text in just seconds. Enjoy a complimentary 15-minute trial without the need for a credit card. By utilizing online automatic subtitling software, you can create subtitles for videos in a fraction of the time it would normally take. Additionally, the platform allows for automatic translation of these subtitles into more than 80 languages. With the Maestra video dubber, you can easily add voiceovers to your videos in foreign languages, utilizing the power of artificial intelligence and synthetic voices to enhance your content's reach and accessibility. This comprehensive solution not only streamlines your workflow but also elevates the quality and versatility of your video productions.

You can use accurate speech recognition at scale and continuously improve model performance by labeling data, training and labeling from one console. We provide state-of the-art speech recognition and understanding at large scale. We do this by offering cutting-edge model training, data-labeling, and flexible deployment options. Our platform recognizes multiple languages and accents. It dynamically adapts to your business' needs with each training session. Enterprise-specific speech transcription software that is fast, accurate, reliable, and scalable. ASR has been reinvented with 100% deep learning, which allows companies to improve their accuracy. Stop waiting for big tech companies to improve their software. Instead, force your developers to manually increase accuracy by using keywords in every API call. You can train your speech model now and reap the benefits in weeks, instead of months or even years.

Subanana is a cutting-edge web application designed for converting audio and video content into subtitles, transcripts, and meeting summaries, supporting over 80 languages with exceptional accuracy, particularly for Asian and mixed-language speech like Cantonese, Mandarin, Japanese, and Korean, which are often inadequately addressed by English-centric tools. Users can easily import files or links from platforms like YouTube, Instagram, or Facebook to create subtitles, which can be customized with a glossary and AI-driven corrections before being exported in various formats such as SRT, VTT, TXT, DOCX, bilingual subtitles, or as burned-in video. For transcripts, the app offers features like speaker identification, the elimination of filler words, and the automatic addition of punctuation and paragraph breaks for clarity. Additionally, it provides templates for meeting summaries that capture decisions and action items, along with a unique bot that integrates with Google Meet and Microsoft Teams to analyze recordings after meetings conclude. Furthermore, Subanana offers live captioning services that provide real-time translations during events, enhancing accessibility and understanding for diverse audiences.

Aiko is an AI transcription app from Sindre Sorhus that helps users convert audio into text on macOS, iOS, and visionOS. The app is powered by OpenAI’s Whisper model and runs transcription locally on the user’s device. This on-device approach makes Aiko well suited for private meetings, lectures, interviews, voice notes, and sensitive recordings. On macOS, Aiko uses the Whisper large v2 model, while iOS uses the medium or small model depending on available device memory. The app supports Shortcuts, giving users flexible ways to record, transcribe, copy results, create subtitles, save text, or connect transcripts to other workflows. Users can transcribe from Finder on macOS, record and transcribe from iPhone shortcuts, or build custom workflows that pass transcripts into apps like Notes or ChatGPT. Aiko includes a free 14-day TestFlight trial with full app access and no auto-charges. Older macOS versions are available for users on macOS 13, 14, and 15. By combining local Whisper transcription, Apple platform support, privacy, and Shortcuts automation, Aiko gives users a simple way to turn speech into text.

AccurateScribe.ai is an advanced cloud-based speech-to-text transcription platform designed to provide fast, highly accurate multilingual transcription services across more than 130 languages and dialects. Leveraging state-of-the-art AI models such as Whisper, it converts audio and video files into precise, readable text with ease and security. The platform accepts a wide range of file formats including MP3, WAV, MP4, and MOV, supporting files as large as 10 hours or 5 GB. Users can also record audio directly through an in-browser voice recorder, which transcribes content in real time, perfect for meetings, lectures, or personal notes. Additionally, AccurateScribe.ai enables transcription from public URLs on platforms like YouTube, Dropbox, and Google Drive without the need for manual file downloads. Its cloud infrastructure ensures fast processing times and secure data handling. The platform caters to a diverse range of transcription needs, from professional and academic to personal use. AccurateScribe.ai simplifies voice-to-text conversion while ensuring flexibility and reliability.

Voice dictation. Transcribe your words on any website in real-time. Echo - Speech-to-Text is an advanced voice typing solution compatible with a wide array of websites. Experience unparalleled accuracy in speech recognition. Notable Features: - ✨ Automatic Punctuation: Benefit from automatic punctuation that ensures your text appears polished and professional. - 🗣️ Direct Voice Typing: Type directly into text fields without dealing with overlays or cumbersome copy-pasting. - 🌍 Support for Multiple Languages: Compatible with over 50 languages, including English, Spanish, German, and French. - 🛠️ Custom Vocabulary Options: Enhance accuracy by adding specialized terms or uncommon words. - ⌨️ Quick Keyboard Shortcuts: Easily start and pause voice recognition using a convenient keyboard shortcut. 🔒 Commitment to Security Your privacy is paramount, as we neither collect nor share your data. We ensure that no dictation text is ever stored in our database. 🛡️ HIPAA Compliance Assured We adhere to HIPAA regulations, ensuring that audio recordings are not retained, and transcription text is securely managed. In addition, our service is designed to provide a seamless and efficient dictation experience, making it an ideal choice for professionals and casual users alike.

The easiest way to record, transcribe, and share your phone's audio right from your smartphone! Trint's mobile application lets you capture the important moments, wherever and whenever you want. Wired: "Amazing!" Google - "Rocket-fueling Innovation!" We know that work doesn't always take place in an office. So we created the mobile app to allow you to access Trint's AI transcription wherever you are. You can record live interviews and import files directly from your phone without any complicated equipment. All you need is the app! Record live conversations. Trint can import audio files from other apps. You can share transcripts and assign editing permissions in-app. Trint transcripts can be easily followed by an intuitive player. All files are saved to your device and to the cloud, so you don't have to worry about losing any. Download audio to your device. While you record, drop markers from your Apple Watch. You can capture in 28 languages right from your iPhone, including English, Spanish and Chinese Mandarin, Hindi, and many more.

Automated transcription service for your audio and video - transcribe and subtitle automatically and accurately. Leverage our feature-rich Online Editor to quickly review and refine your transcript. Collaborate, share and export your transcript with your audience or your team. Start your free trial now, no credit card required. Prices start at $6 per audio our (purchased transcription credit never expire)

The NVIDIA Nemotron 3 Nano Omni represents a groundbreaking open foundation model that integrates various modes of perception and reasoning—including text, images, audio, video, and documents—into a single streamlined architecture. By eliminating the necessity for distinct models tailored to each modality, it effectively minimizes inference delays, simplifies orchestration, and lowers costs while ensuring a cohesive cross-modal context. This innovative model is specifically engineered for agentic AI systems, functioning as a perception and context sub-agent that empowers larger AI entities to perceive and interpret their surroundings in real-time across various formats such as screens, recordings, and both structured and unstructured data. Its capabilities extend to complex multimodal reasoning tasks, encompassing document comprehension, speech recognition, extensive audio-video analysis, and intricate computer workflows, thus allowing agents to navigate dynamic interfaces and multifaceted environments with ease. With a hybrid architecture that is finely tuned for handling long contexts and high throughput, the Nemotron 3 Nano Omni is adept at managing sizable inputs, including multi-page documents, making it a versatile tool in the realm of AI development. Not only does it unify modalities, but it also enhances the overall efficiency of intelligent systems in processing and understanding diverse data types.

Mistral AI has introduced Voxtral Transcribe 2, an advanced suite of speech-to-text models that provides remarkably fast, high-quality audio transcription and speaker identification, supporting a diverse range of languages. This collection features Voxtral Mini Transcribe V2, which is tailored for batch transcription and includes functionalities like word-level timestamps, context biasing, and compatibility with 13 different languages, alongside Voxtral Realtime, which is optimized for live speech recognition with adjustable latency that can drop below 200 ms for immediate use cases. Both models excel in transcription accuracy while maintaining efficiency and cost-effectiveness; Mini Transcribe V2 is noted for its exceptional performance and minimal error rates, while Realtime is made available as open-source under the Apache 2.0 license, enabling developers to implement it on edge devices or within secure environments. Furthermore, the innovative technology embedded in these models represents a significant leap forward in transcription solutions, catering to various applications across industries.

ElevenLabs has unveiled Scribe, a cutting-edge Automatic Speech Recognition (ASR) model that aims to provide remarkably accurate transcriptions in 99 different languages. This innovative system is tailored to effectively manage a wide range of real-world audio situations, featuring capabilities such as word-level timestamps, speaker identification, and audio-event tagging. In benchmark evaluations like FLEURS and Common Voice, Scribe has outperformed leading models, including Gemini 2.0 Flash, Whisper Large V3, and Deepgram Nova-3, achieving impressive word error rates of 98.7% for Italian and 96.7% for English. Additionally, Scribe shows a significant reduction in errors for languages that have often faced challenges, such as Serbian, Cantonese, and Malayalam, where competing models frequently report error rates above 40%. Furthermore, developers can easily incorporate Scribe into their applications via ElevenLabs' speech-to-text API, which returns structured JSON transcripts enriched with comprehensive annotations. This level of accessibility and performance is set to revolutionize the field of transcription and enhance the user experience across various applications.

Beey is a highly efficient application that transforms audio and video files into text within minutes, boasting remarkable accuracy. It supports speech recognition in 20 different languages, making it versatile for a global audience. Additionally, its intuitive editing tool allows users to refine the transcribed content, export it in multiple formats, and generate automatic subtitles or translations. The editing interface features a synchronized playback preview that aligns with the edited text, highlighted by a moving cursor, enabling seamless adjustments. Users can control the playback speed, slow it down, speed it up, or start from any chosen point in the transcription. Furthermore, Beey encompasses a range of supplementary tools: Link, Splitter, Stream, and Voice. The Link tool enables direct transcription of audio or video from major platforms like YouTube. The Splitter feature is particularly useful for lengthy recordings, breaking them into manageable segments for individual editing. Stream allows for real-time transcription and captioning of live broadcasts, while the Voice tool is designed for recording and transcribing live speech effortlessly. Overall, Beey provides a comprehensive suite of features that enhance the transcription experience, catering to various user needs.

Speechyou is an innovative web and iOS application that converts recorded speech along with uploaded audio or video files into searchable and timestamped text formats. Users have the capability to record voice notes or meetings directly, as well as import existing media files, allowing them to review transcripts, recognize speakers, and export their text or subtitle files easily. The app also features Ask AI, which can respond to inquiries related to transcripts and produce summaries, highlight key points, and outline action items; in addition, it provides translation options for transcripts, facilitates note organization, and allows for sharing of content. Tailored for both business teams and individual professionals, Speechyou serves a diverse range of users including journalists, researchers, students, and educators engaged in activities such as meetings, interviews, lectures, voice memos, podcasts, and other audio or video recordings. Its user-friendly browser interface, along with dedicated iPhone and iPad applications, enhances the overall experience by supporting seamless workflows for recording, transcription, searching, and reviewing. This versatility makes it an invaluable tool for anyone looking to streamline their audio and video content management.

You can upload any audio or video file, as we support all formats. After uploading, you can check your transcript, which includes timestamps and identifies speakers. The transcripts are available for saving and exporting in various formats such as MS Word, PDF, SRT, VTT, and more. The accuracy of the transcript is influenced by the quality of the audio, so ensure that your recordings are clear for the best results. With Temi's complimentary transcription editor, you can make quick edits to your transcripts online in just minutes. This tool is developed by experts in machine learning and speech recognition. You can easily refine the generated transcript, modify playback speed, and navigate through the content swiftly. Temi tracks the timing of each word meticulously, allowing you to add specific timestamps. Each change in speaker is marked and labeled for clarity. Finally, you can download your transcript in text formats like MS Word or PDF, or as closed caption files in SRT or VTT formats for your convenience. This comprehensive service ensures that you have all the tools necessary for effective transcription management.

MAI-Transcribe-1.5 represents Microsoft AI’s advanced speech-to-text solution, expertly converting challenging audio into precise, contextually relevant transcripts in 43 different languages. This model ensures reliable and high-accuracy transcription that accommodates various languages, accents, speaking styles, and difficult audio environments, incorporating automatic language detection for added convenience. It is expertly crafted to handle real-world audio scenarios, such as those found in conference rooms, over phone calls, in bustling streets, and even from low-quality recordings that might include background noise or overlapping dialogue. Furthermore, MAI-Transcribe-1.5 is tailored to understand and utilize domain-specific language, making it incredibly useful for tasks like captioning, call analysis, enhancing accessibility, transcribing meetings, recording doctor’s notes, managing pharma customer interactions, and streamlining content workflows, all without requiring extensive setup. The model leverages contextual biasing to enhance its comprehension of specialized vocabulary, names, and industry-specific jargon that standard transcription systems often overlook, ensuring that users receive the most accurate and relevant transcripts possible. By seamlessly integrating into various enterprise applications, it significantly enhances productivity and communication efficiency in professional settings.

Palatine Speech serves as a cloud-based platform and API provider specializing in AI-driven speech processing solutions. It offers a wide array of features, including transcription, speaker diarization, word timestamps, automatic language detection, translation capabilities, SRT/VTT subtitle generation, sentiment analysis, and text summarization. The API is versatile, accommodating both streaming and asynchronous processing, alongside custom dictionaries and OpenAI-compatible endpoints, supporting over 100 languages and more than 23 audio and video formats. Users can choose between cloud and on-premise deployment options. Additionally, Palatine is the creator of Palatine Murmur 0.4.0, a privacy-focused application designed for meeting recording, transcription, and AI-powered summarization, compatible with macOS, Windows, and Linux systems, ensuring users have a comprehensive toolset for managing their audio and video needs. This application underscores Palatine's commitment to enhancing user privacy while delivering advanced functionality.

NVIDIA has created the Nemotron family of open-source models aimed at producing synthetic data specifically for training large language models (LLMs) intended for commercial use. Among these, the Nemotron-4 340B model stands out as a key innovation, providing developers with a robust resource to generate superior quality data while also allowing for the filtering of this data according to multiple attributes through a reward model. This advancement not only enhances data generation capabilities but also streamlines the process of training LLMs, making it more efficient and tailored to specific needs.

The automatic speech recognition (ASR) system developed by GoVivace accommodates a variety of English accents and is adaptable to numerous languages, making it versatile for global use. Additionally, this ASR technology is compatible with standard telephony, as well as web and mobile platforms. It efficiently executes voice commands issued to devices such as computers, tablets, smartphones, and telephones, utilizing a microphone for input, which allows for a wide range of applications. The GoVivace ASR engine works by comparing spoken input to an array of predetermined options, converting the verbal communication into text. This array of predetermined options forms the grammar for the application, serving as the critical link between the speaker and the underlying processing system. Remarkably, GoVivace's innovative speech recognition solution operates effectively with minimal grammar requirements, yet it is robust enough to handle extensive grammars for more intricate tasks, showcasing its flexibility and efficiency. Such adaptability makes it suitable for various industries and user needs, further broadening its market appeal.