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speech-to-text Checked

Transcribe audio to text using ElevenLabs Scribe v2. Use when converting audio/video to text, gen…
✍️ Writing skills By calesthio Version v1.0.0 Added 2026-09-07 Updated 2026-09-07 Source ↗
8.4Overall rating
2Views
5Downloads
56.5kStars

Installation

🤖 Install via AI

Copy the prompt below and send it to your AI assistant (e.g. Claude Code) — it will follow the instructions and install automatically.

Install the "speech-to-text" skill by following the instructions at https://skill123.me/install/speech-to-text.
⌨️ Command line install

Run in your terminal — downloads and installs to ~/.claude/skills/.

curl -fsSL https://skill123.me/install/speech-to-text.sh | bash
📦 Download ZIP

Download the zip and extract it into your skills directory (e.g. ~/.claude/skills/), then restart your session.

⬇ Download v1.0.0 · 16 KB
📊 Static rating
8.4/10
Trigger
7.0
Structure
8.7
Workflow
8.0
Content
9.0
Engineering
9.0
Security
9.5

Six-dimension rubric with reviewer rationale.

View scorecard ↓
🛡 Dynamic test
—

Not live-tested yet — this skill is rated on static analysis only.

About this skill

Transcribe audio to text using ElevenLabs Scribe v2. Use when converting audio/video to text, generating subtitles, transcribing meetings, or processing spoken content.

ElevenLabs Speech-to-Text

Transcribe audio to text with Scribe v2 - supports 90+ languages, speaker diarization, and word-level timestamps.

> **Setup:** See Installation Guide (references/installation.md). For JavaScript, use @elevenlabs/* packages only.

Documentation

Transcribe audio to text using ElevenLabs Scribe v2. Use when converting audio/video to text, generating subtitles, transcribing meetings, or processing spoken content.

ElevenLabs Speech-to-Text

Transcribe audio to text with Scribe v2 - supports 90+ languages, speaker diarization, and word-level timestamps.

> **Setup:** See Installation Guide (references/installation.md). For JavaScript, use @elevenlabs/* packages only.

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 |

|----------|-------------|----------|

| scribe_v2 | State-of-the-art accuracy, 90+ languages | Batch transcription, subtitles, long-form audio |

| scribe_v2_realtime | 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

View full documentation

{
  "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:**

  • word - An actual spoken word
  • spacing - Whitespace between words (useful for precise timing)
  • audio_event - Non-speech sounds the model detected (laughter, applause, music, etc.)

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 commit() 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 |

|-------|-------------|

| partial_transcript | Live interim results |

| committed_transcript | Final results after commit |

| committed_transcript_with_timestamps | Final with word timing |

| error | Error occurred |

See real-time references for complete documentation.

References

  • Installation Guide (references/installation.md)
  • Transcription Options (references/transcription-options.md)
  • Real-Time Client-Side Streaming (references/realtime-client-side.md)
  • Real-Time Server-Side Streaming (references/realtime-server-side.md)
  • Commit Strategies (references/realtime-commit-strategies.md)
  • Real-Time Event Reference (references/realtime-events.md)

Score breakdown

Trigger
7.0
Clear What/When with subtitle-meeting-transcribe scenarios, but no When-Not boundary
Structure
8.7
275-line SKILL.md with six topical references properly offloaded, no redundant files
Workflow
8.0
Model-choice table, error codes with handling code, commit strategies; weak output verification loop
Content
9.0
Tight imperative sections, complete runnable snippets, consistent tables and code blocks
Engineering
9.0
Valid frontmatter with env metadata, all six referenced files exist, packages declared
Security
9.5
Only official elevenlabs.io endpoints, env-var API key, base64 is audio payload not obfuscation
Well-built ElevenLabs Scribe v2 wrapper: concise SKILL.md with six purposeful references, runnable Python/JS/curl examples, error codes, and realtime commit-strategy guidance; lacks When-Not and output verification.