Speechmatics
Best-in-Market Speech-to-Text & Voice AI for Enterprises.
Speechmatics delivers industry-leading Speech-to-Text and Voice AI for enterprises needing unrivaled accuracy, security, and flexibility. Our enterprise-grade APIs provide real-time and batch transcription with exceptional precision—across the widest range of languages, dialects, and accents.
Powered by Foundational Speech Technology, Speechmatics supports mission-critical voice applications in media, contact centers, finance, healthcare, and more. With on-prem, cloud, and hybrid deployment, businesses maintain full control over data security while unlocking voice insights.
Trusted by global leaders, Speechmatics is the top choice for best-in-class transcription and voice intelligence.
🔹 Unmatched Accuracy – Superior transcription across languages & accents
🔹 Flexible Deployment – Cloud, on-prem, and hybrid
🔹 Enterprise-Grade Security – Full data control
🔹 Real-Time & Batch Processing – Scalable transcription
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Scribe
ElevenLabs has introduced Scribe, an advanced Automatic Speech Recognition (ASR) model designed to deliver highly accurate transcriptions across 99 languages. Scribe is engineered to handle diverse real-world audio scenarios, providing features such as word-level timestamps, speaker diarization, and audio-event tagging. Benchmark tests, including FLEURS and Common Voice, demonstrate Scribe's superior performance over leading models like Gemini 2.0 Flash, Whisper Large V3, and Deepgram Nova-3, achieving the lowest word error rates in languages such as Italian (98.7%) and English (96.7%). Notably, Scribe also significantly reduces errors in languages that have been traditionally underserved, including Serbian, Cantonese, and Malayalam, where other models often exhibit error rates exceeding 40%. Developers can integrate Scribe through ElevenLabs' speech-to-text API, receiving structured JSON transcripts that include detailed annotations.
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Gladia
Gladia is an advanced audio transcription and intelligence platform delivered via a unified API that supports both asynchronous (pre-recorded) and real-time streaming transcription, enabling developers to convert speech to text in over 100 languages with features like word-level timestamps, language detection, code-switching, speaker diarization, translation, summarization, custom vocabulary, and entity extraction. Its real-time engine achieves latencies under 300 ms while maintaining high accuracy, and it offers “partials” (intermediate transcripts) to improve responsiveness in live settings. The platform’s asynchronous API is powered by a proprietary Whisper-Zero model optimized for enterprise audio, and it lets clients apply add-ons such as enhanced punctuation, name consistency, custom metadata tagging, and export to subtitle formats (SRT, VTT).
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Amazon Transcribe
Amazon Transcribe makes it easy for developers to add speech to text capabilities to their applications. Audio data is virtually impossible for computers to search and analyze. Therefore, recorded speech needs to be converted to text before it can be used in applications. Historically, customers had to work with transcription providers that required them to sign expensive contracts and were hard to integrate into their technology stacks to accomplish this task. Many of these providers use outdated technology that does not adapt well to different scenarios, like low-fidelity phone audio common in contact centers, which results in poor accuracy. Amazon Transcribe uses a deep learning process called automatic speech recognition (ASR) to convert speech to text quickly and accurately. Amazon Transcribe can be used to transcribe customer service calls, automate subtitling, and generate metadata for media assets to create a fully searchable archive.
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