Benefits of Live Transcription API
Real-time transcription API enables developers to add instant speech-to-text to applications. Stream audio and receive transcribed text with minimal latency.
Key capabilities include:
- Sub-second transcription latency
- WebSocket streaming support
- 50+ language support
- Speaker diarization
- Punctuation and formatting
Build live captioning, voice commands, and accessibility features with reliable transcription.
How Real-Time API Works
- Establish WebSocket connection
- Stream audio in supported format
- Receive transcription results in real-time
- Process partial and final results
- Handle speaker changes and formatting
API documentation includes code examples for major programming languages and frameworks.
Who Needs Transcription API
Real-time transcription API serves developers:
- App developers adding voice features
- Accessibility teams building live captions
- Call center platforms transcribing support calls
- Meeting apps providing live transcription
- Voice assistant developers processing commands
- Broadcast platforms generating live subtitles
Any application needing live speech-to-text benefits from transcription API.
FAQ
What is real-time transcription API latency?
Quality APIs deliver results within 200-500 milliseconds of speech, enabling live captioning and responsive voice applications.
What audio formats does the API accept?
Most APIs accept PCM, WAV, MP3, and FLAC formats. WebSocket streaming typically uses raw PCM for lowest latency.
How accurate is live transcription?
Real-time accuracy typically reaches 90-95% for clear speech. Accuracy improves with domain-specific vocabulary customization.
Does the API support speaker identification?
Yes, speaker diarization identifies different speakers in audio streams, useful for multi-party conversations and meetings.
What are API pricing models?
Pricing typically charges per audio minute processed. Volume discounts available for high-usage applications.
