OpenMontage speech-to-text
Transcribe audio to text using ElevenLabs Scribe v2. Use when converting audio/video to text, generating subtitles, transcribing meetings, or processing spoken content.
git clone https://github.com/calesthio/OpenMontage
T=$(mktemp -d) && git clone --depth=1 https://github.com/calesthio/OpenMontage "$T" && mkdir -p ~/.claude/skills && cp -r "$T/.claude/skills/speech-to-text" ~/.claude/skills/calesthio-openmontage-speech-to-text-e8f862 && rm -rf "$T"
.claude/skills/speech-to-text/SKILL.mdElevenLabs Speech-to-Text
Transcribe audio to text with Scribe v2 - supports 90+ languages, speaker diarization, and word-level timestamps.
Setup: See Installation Guide. For JavaScript, use
packages only.@elevenlabs/*
Quick Start
Python
from elevenlabs import ElevenLabs client = ElevenLabs() with open("audio.mp3", "rb") as audio_file: result = client.speech_to_text.convert(file=audio_file, model_id="scribe_v2") print(result.text)
JavaScript
import { ElevenLabsClient } from "@elevenlabs/elevenlabs-js"; import { createReadStream } from "fs"; const client = new ElevenLabsClient(); const result = await client.speechToText.convert({ file: createReadStream("audio.mp3"), modelId: "scribe_v2", }); console.log(result.text);
cURL
curl -X POST "https://api.elevenlabs.io/v1/speech-to-text" \ -H "xi-api-key: $ELEVENLABS_API_KEY" -F "file=@audio.mp3" -F "model_id=scribe_v2"
Models
| Model ID | Description | Best For |
|---|---|---|
| State-of-the-art accuracy, 90+ languages | Batch transcription, subtitles, long-form audio |
| Low latency (~150ms) | Live transcription, voice agents |
Transcription with Timestamps
Word-level timestamps include type classification and speaker identification:
result = client.speech_to_text.convert( file=audio_file, model_id="scribe_v2", timestamps_granularity="word" ) for word in result.words: print(f"{word.text}: {word.start}s - {word.end}s (type: {word.type})")
Speaker Diarization
Identify WHO said WHAT - the model labels each word with a speaker ID, useful for meetings, interviews, or any multi-speaker audio:
result = client.speech_to_text.convert( file=audio_file, model_id="scribe_v2", diarize=True ) for word in result.words: print(f"[{word.speaker_id}] {word.text}")
Keyterm Prompting
Help the model recognize specific words it might otherwise mishear - product names, technical jargon, or unusual spellings (up to 100 terms):
result = client.speech_to_text.convert( file=audio_file, model_id="scribe_v2", keyterms=["ElevenLabs", "Scribe", "API"] )
Language Detection
Automatic detection with optional language hint:
result = client.speech_to_text.convert( file=audio_file, model_id="scribe_v2", language_code="eng" # ISO 639-1 or ISO 639-3 code ) print(f"Detected: {result.language_code} ({result.language_probability:.0%})")
Supported Formats
Audio: MP3, WAV, M4A, FLAC, OGG, WebM, AAC, AIFF, Opus Video: MP4, AVI, MKV, MOV, WMV, FLV, WebM, MPEG, 3GPP
Limits: Up to 3GB file size, 10 hours duration
Response Format
{ "text": "The full transcription text", "language_code": "eng", "language_probability": 0.98, "words": [ {"text": "The", "start": 0.0, "end": 0.15, "type": "word", "speaker_id": "speaker_0"}, {"text": " ", "start": 0.15, "end": 0.16, "type": "spacing", "speaker_id": "speaker_0"} ] }
Word types:
- An actual spoken wordword
- Whitespace between words (useful for precise timing)spacing
- Non-speech sounds the model detected (laughter, applause, music, etc.)audio_event
Error Handling
try: result = client.speech_to_text.convert(file=audio_file, model_id="scribe_v2") except Exception as e: print(f"Transcription failed: {e}")
Common errors:
- 401: Invalid API key
- 422: Invalid parameters
- 429: Rate limit exceeded
Tracking Costs
Monitor usage via
request-id response header:
response = client.speech_to_text.convert.with_raw_response(file=audio_file, model_id="scribe_v2") result = response.parse() print(f"Request ID: {response.headers.get('request-id')}")
Real-Time Streaming
For live transcription with ultra-low latency (~150ms), use the real-time API. The real-time API produces two types of transcripts:
- Partial transcripts: Interim results that update frequently as audio is processed - use these for live feedback (e.g., showing text as the user speaks)
- Committed transcripts: Final, stable results after you "commit" - use these as the source of truth for your application
A "commit" tells the model to finalize the current segment. You can commit manually (e.g., when the user pauses) or use Voice Activity Detection (VAD) to auto-commit on silence.
Python (Server-Side)
import asyncio from elevenlabs import ElevenLabs client = ElevenLabs() async def transcribe_realtime(): async with client.speech_to_text.realtime.connect( model_id="scribe_v2_realtime", include_timestamps=True, ) as connection: await connection.stream_url("https://example.com/audio.mp3") async for event in connection: if event.type == "partial_transcript": print(f"Partial: {event.text}") elif event.type == "committed_transcript": print(f"Final: {event.text}") asyncio.run(transcribe_realtime())
JavaScript (Client-Side with React)
import { useScribe, CommitStrategy } from "@elevenlabs/react"; function TranscriptionComponent() { const [transcript, setTranscript] = useState(""); const scribe = useScribe({ modelId: "scribe_v2_realtime", commitStrategy: CommitStrategy.VAD, // Auto-commit on silence for mic input onPartialTranscript: (data) => console.log("Partial:", data.text), onCommittedTranscript: (data) => setTranscript((prev) => prev + data.text), }); const start = async () => { // Get token from your backend (never expose API key to client) const { token } = await fetch("/scribe-token").then((r) => r.json()); await scribe.connect({ token, microphone: { echoCancellation: true, noiseSuppression: true }, }); }; return <button onClick={start}>Start Recording</button>; }
Commit Strategies
| Strategy | Description |
|---|---|
| Manual | You call when ready - use for file processing or when you control the audio segments |
| VAD | Voice Activity Detection auto-commits when silence is detected - use for live microphone input |
// React: set commitStrategy on the hook (recommended for mic input) import { useScribe, CommitStrategy } from "@elevenlabs/react"; const scribe = useScribe({ modelId: "scribe_v2_realtime", commitStrategy: CommitStrategy.VAD, // Optional VAD tuning: vadSilenceThresholdSecs: 1.5, vadThreshold: 0.4, });
// JavaScript client: pass vad config on connect const connection = await client.speechToText.realtime.connect({ modelId: "scribe_v2_realtime", vad: { silenceThresholdSecs: 1.5, threshold: 0.4, }, });
Event Types
| Event | Description |
|---|---|
| Live interim results |
| Final results after commit |
| Final with word timing |
| Error occurred |
See real-time references for complete documentation.