acestep
AI music generation with ACE-Step 1.5 — background music, vocal tracks, covers, stem extraction for video production. Use when…
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
Claude Code
$ npx -y skills add openmontageinc/openmontage --skill azure-speech-to-text --agent claude-code
Codex
$ npx skills add openmontageinc/openmontage --skill azure-speech-to-text -a codex -y
Cursor
$ npx skills add openmontageinc/openmontage --skill azure-speech-to-text -a cursor -y
Gemini CLI
$ npx skills add openmontageinc/openmontage --skill azure-speech-to-text -a gemini-cli -y
GitHub Copilot
$ npx skills add openmontageinc/openmontage --skill azure-speech-to-text -a github-copilot -y
OpenCode
$ npx skills add openmontageinc/openmontage --skill azure-speech-to-text -a opencode -y
Windsurf
$ npx skills add openmontageinc/openmontage --skill azure-speech-to-text -a windsurf -y
name: azure-speech-to-text
description: 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.
license: MIT
compatibility: Requires internet access and an Azure AI Speech resource (AZURE_SPEECH_KEY + AZURE_SPEECH_REGION).
metadata: {"openclaw": {"requires": {"env": ["AZURE_SPEECH_KEY", "AZURE_SPEECH_REGION"]}, "primaryEnv": "AZURE_SPEECH_KEY"}}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](https://learn.microsoft.com/azure/ai-services/speech-service/fast-transcription-create) · [Speech service overview](https://learn.microsoft.com/azure/ai-services/speech-service/spx-overview)
Azure exposes three STT surfaces. OpenMontage uses **Fast Transcription** because the pipeline transcribes **local audio files**:
| Surface | Input | Latency | Needs | |---------|-------|---------|-------| | **Fast Transcription** (used here) | local file, multipart POST | synchronous, sub-real-time | key + region | | Batch Transcription | audio at a URL (Blob + SAS) | async job + polling | Blob storage plumbing | | Speech SDK (`spx`) | mic / stream / file | streaming | native `azure-cognitiveservices-speech` package |
Fast Transcription needs no Blob storage, no SAS URLs, and no native SDK — just `requests` and the two env vars.
Create a **Speech** resource in the [Azure portal](https://portal.azure.com); 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.
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.
when you know the language; it is faster and more accurate than auto-ID.
identification across this shortlist. Narrow it to the languages you actually expect; a huge list slows detection and invites misclassification.
podcasts). Set `max_speakers` to the real upper bound.
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`.
jobs, use Azure Batch Transcription instead.
you only need speech — smaller upload, same result.
the first and last cues against the source audio.
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