
Fathom is an AI meeting assistant that helps users capture, summarize, search, and act on meetings with less manual work. The platform creates accurate transcripts, instant summaries, action items, and follow-up notes so users can focus on live conversations instead of taking notes. Fathom supports both traditional meeting capture and bot-free capture through its desktop app. Teams can use Fathom as a shared source of truth across customer calls, internal meetings, strategy sessions, and project conversations. Ask Fathom lets users search across meetings and ask questions about conversations, decisions, commitments, risks, and next steps. The platform also supports topic monitoring so important moments and signals are easier to find. Fathom syncs meeting notes, insights, and action items into tools such as Slack, Salesforce, HubSpot, Notion, Asana, Gmail, Zoom, Google Meet, Microsoft Teams, ChatGPT, Claude, Zapier, and API or MCP workflows. It supports security and compliance needs with SOC 2 Type II, GDPR, HIPAA compliance, SSO, and SCIM. By combining AI notetaking, bot-free capture, transcripts, summaries, integrations, search, and workflow automation, Fathom helps teams move from meetings to execution faster.
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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.
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Muse Voice Transcribe
Muse Voice Transcribe represents Meta’s inaugural venture into real-time audio perception, providing instantaneous automatic speech recognition (ASR), speaker diarization, and endpointing capabilities. This autoregressive multimodal model, part of the Muse Spark series, analyzes audio segments of 80 milliseconds and makes real-time decisions on whether to keep listening or to convert the spoken words into text. The adaptive delay mechanism allows it to adjust the audio context utilized for each word according to the complexity of the speech, thus optimizing the balance between transcription precision and response time. With training encompassing over 70 languages, 25 of which were rigorously validated at the time of its release, the model also seamlessly accommodates arbitrary code-switching, allowing transitions within and across sentences. Furthermore, language, keyword, and contextual biasing features enhance the recognition capabilities for specific names, locations, contacts, or specialized terms. The streaming diarization functionality enables the model to recognize shifts in speakers and can differentiate between more than 20 individual voices. Additionally, the endpointing feature is adept at identifying the commencement of speech and knowing when a user has completed their statement, ensuring a fluid interaction experience. Overall, Muse Voice Transcribe stands out as a cutting-edge tool in the realm of speech recognition technology, merging advanced features with user-friendly application.
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Nemotron 3.5 Transcribe
NVIDIA Nemotron 3.5 Transcribe serves as an open-weight model tailored for developers aiming to create transcription and audio-processing systems. This model transforms audio recordings into text, offering options for both verbatim and edited transcripts, along with features like contextual vocabulary, speaker identification, word-level timestamps, speech translation, language detection, and audio question answering. It boasts a language matrix that encompasses 40 different languages, though the extent of task coverage may vary from one language to another. Capable of processing recordings of up to one hour, the model is optimized for offline inference on compatible NVIDIA GPU hardware and supported runtimes. Developers can seamlessly integrate it into a variety of applications, including speech interfaces, media and meeting processing, customer service analytics, translation software, and other audio comprehension tools. Furthermore, the model's weights, alongside a fine-tuning toolkit, are made available under the OpenMDW-1.1 license, with specific hardware and serving requirements contingent upon the nature of the deployment. This flexibility allows teams to adapt the model for diverse use cases in the field of audio processing and transcription.
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