AssemblyAI
Automatically convert audio and video files and live audio streams to text with AssemblyAI's speech-to-text APIs. Do more with audio intelligence, summarization, content moderation, topic detection, and more. Powered by cutting-edge AI models. From in-depth tutorials to detailed changelogs, to comprehensive documentation, AssemblyAI is focused on providing developers a great experience every step of the way. From core speech-to-text conversion to sentiment analysis, our simple API offers a full suite of solutions catered to all your business speech-to-text needs. We work with startups of all sizes, from early-stage startups to scale-ups, by providing cost-efficient speech-to-text solutions. We're built for scale. We process millions of audio files every day for hundreds of customers, including dozens of Fortune 500 enterprises.
Universal-2: Our most advanced speech-to-text model captures the complexity of human speech for impeccable audio data that powers sharper insights.
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Voxtral Mini Transcribe Realtime 3
Voxtral Mini Transcribe Realtime 3 is a speech-to-text model delivered through Mistral’s API for applications that need live transcription. It streams audio over WebSocket and returns transcript events while a person is speaking, with configurable streaming delays. Developers can use it to add real-time captions, voice-assistant input, or transcription to voice-agent workflows. The model provides turn detection, speaker diarization, and context biasing for domain-specific vocabulary, and supports transcription across 45 languages and dialects. Applications can connect to its real-time endpoint and receive transcription events as audio arrives, rather than waiting for an entire recording to finish. Voxtral Mini Transcribe Realtime 3 is part of Mistral’s Voxtral audio model family.
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Seeduplex
Seeduplex is a native full-duplex speech large language model built on a new “listen while speaking” framework for more natural, fluid, and precisely paced voice interaction. Unlike traditional half-duplex systems that alternate between listening and replying, it continuously receives and understands user-side audio, allowing it to listen and speak simultaneously while tracking the broader acoustic environment. Its high-precision interference suppression distinguishes genuine user interaction from background noise, broadcasts, navigation prompts, side conversations, and overlapping voices, reducing false responses and false interruptions in complex settings. Seeduplex also combines speech and semantic features for adaptive endpoint detection, helping it recognize when a user is thinking, hesitating, correcting themselves, or has actually finished speaking. It can wait patiently through reflective pauses, respond quickly once an utterance ends, and stop smoothly when interrupted.
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Azure Voice Live API
Azure Voice Live API is a fully managed solution for building low-latency, high-quality speech-to-speech agents through one unified interface. It combines speech recognition, generative AI, and text-to-speech, allowing developers to send audio input and receive audio output, synchronized avatar visuals, and action triggers without manually orchestrating separate backend components or deploying the underlying models. It supports more than 140 speech-to-text locales and over 600 standard voices across 150+ text-to-speech locales, with options for phrase lists, custom speech, custom voices, and brand-aligned avatars. Developers can choose among multiple generative AI models, including GPT-Realtime, GPT-5, GPT-4.1, GPT-4o, Phi, and compatible bring-your-own models, depending on the intelligence, speed, and latency required. Advanced conversational features include noise suppression, echo cancellation, robust interruption detection, and end-of-turn detection.
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