calesthio/openmontageMIT1 file

Azure Speech To Text

Transcribe audio to text using Azure AI Speech (Fast Transcription REST API). Use when converting audio/video to text, generating subtitles, or processing spoken content in OpenMontage. Optional cloud STT provider — preferred when AZURE_SPEECH_KEY is configured; the local faster-whisper `transcriber` is the default offline path.

Specification
Skill ID
calesthio/openmontage/azure-speech-to-text
Publisher
calesthio
Repository
openmontage
Installs
0
Files
1
License
MIT
Requires
Requires internet access and an Azure AI Speech resource (AZURE_SPEECH_KEY + AZURE_SPEECH_REGION).
Synced
Sep 23, 2026
How to use it

Open any RiverX project, open the Skills panel in the chat, and search for this identifier. The files are fetched from the source repository at install time.

calesthio/openmontage/azure-speech-to-textInstalls these files
  • SKILL.md

What this skill tells the agent

Azure AI Speech — Speech-to-Text

Transcribe audio to text with Azure Fast Transcription — synchronous, word-level timestamps, speaker diarization, and multi-language identification. In OpenMontage this is exposed through the azure_stt tool (capability=analysis, provider=azure). It is an optional cloud STT provider — when AZURE_SPEECH_KEY is configured, prefer it for cloud transcription. The local transcriber tool (faster-whisper) remains the default offline path and the fallback when Azure is unavailable.

Docs: Fast Transcription · Speech service overview

Why Fast Transcription (not Batch)

Azure exposes three STT surfaces. OpenMontage uses Fast Transcription because the pipeline transcribes local audio files:

SurfaceInputLatencyNeeds
Fast Transcription (used here)local file, multipart POSTsynchronous, sub-real-timekey + region
Batch Transcriptionaudio at a URL (Blob + SAS)async job + pollingBlob storage plumbing
Speech SDK (spx)mic / stream / filestreamingnative azure-cognitiveservices-speech package

Fast Transcription needs no Blob storage, no SAS URLs, and no native SDK — just requests and the two env vars.

Setup

Create a Speech resource in the Azure portal; copy the key and region from its Keys and Endpoint page.

export AZURE_SPEECH_KEY=your_speech_resource_key
export AZURE_SPEECH_REGION=eastus          # your resource's region
# export AZURE_SPEECH_ENDPOINT=https://...  # optional: overrides region

azure_stt reports AVAILABLE once AZURE_SPEECH_KEY plus either AZURE_SPEECH_REGION or AZURE_SPEECH_ENDPOINT are set.

Using it in a pipeline

Prefer azure_stt over transcriber unless the run must be offline. Its output matches the transcriber schema exactly, so it is a drop-in for subtitle_gen and any stage that consumes a transcript.

from tools.tool_registry import registry
registry.discover()
stt = registry._tools["azure_stt"]

result = stt.execute({
    "input_path": "projects/my-video/assets/audio/narration.mp3",
    # "language": "en",          # ISO 639-1 or BCP-47 ("en-US"); omit for auto-ID
    # "diarize": True,           # speaker labels, no HuggingFace token needed
    # "max_speakers": 4,
    "output_dir": "projects/my-video/artifacts",
})
if result.success:
    segs = result.data["segments"]          # [{id,start,end,text,words:[...]}]
    words = result.data["word_timestamps"]  # flat [{word,start,end,probability}]

If azure_stt is unavailable (no key) or errors, fall back to transcriber (local whisper) — its execute signature and output are identical.

Parameters that matter

  • `language` — pass an ISO code ("en") or a full locale ("en-US"). Pin it when you know the language; it is faster and more accurate than auto-ID.
  • `candidate_locales` — when language is omitted, Azure runs language identification across this shortlist. Narrow it to the languages you actually expect; a huge list slows detection and invites misclassification.
  • `diarize` / `max_speakers` — enable for multi-speaker audio (interviews, podcasts). Set max_speakers to the real upper bound.
  • `profanity_filter` — None | Masked (default) | Removed | Tags.

Response shape (mapped to the transcriber schema)

The raw Azure response (phrases[] with offsetMilliseconds / words[]) is converted to seconds and the OpenMontage transcript schema:

{
  "segments": [
    {"id": 0, "start": 0.0, "end": 2.4, "text": "Hello world",
     "speaker": 1,
     "words": [{"word": "Hello", "start": 0.0, "end": 0.5, "probability": 0.98}]}
  ],
  "word_timestamps": [{"word": "Hello", "start": 0.0, "end": 0.5, "probability": 0.98}],
  "language": "en-US",
  "duration_seconds": 2.4,
  "provider": "azure"
}

Note: Fast Transcription has no per-word confidence, so each word carries the phrase confidence in probability.

Limits & tips

  • Single file up to ~2 hours / a few hundred MB per request. For longer or bulk jobs, use Azure Batch Transcription instead.
  • Send clean audio (16 kHz+ mono is plenty). Transcode video to audio first if you only need speech — smaller upload, same result.
  • Verify timing: word timestamps drive subtitle cues in subtitle_gen. Spot-check the first and last cues against the source audio.

Use this skill in a project

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