stepfun-asr
Transcribe audio with StepFun's stepaudio-2.5-asr — an SSE endpoint (NOT /v1/audio/transcriptions) with 32K context, ~85-101x RTF on long audio, and a single-call ceiling around 30 minutes (no client-side chunking). Use when transcribing Chinese / English audio with StepFun, when long-form recordings (5-30 min) need to land in one request, when migrating from step-asr / step-asr-1.1, or when hitting the misleading `model stepaudio-2.5-asr not supported` error (which actually means wrong endpoint). Triggers on 阶跃 ASR, StepFun ASR, stepaudio-2.5-asr, 转录, 语音识别, 长音频转写, 语音转文字. For TTS with the sibling stepaudio-2.5-tts model, use the stepfun-tts skill instead.
How do I install this agent skill?
npx skills add https://github.com/daymade/claude-code-skills --skill stepfun-asrIs this agent skill safe to install?
- Gen Agent Trust Hubpass
This skill provides tools and documentation to transcribe audio using StepFun's stepaudio-2.5-asr model. It includes a Python script that correctly handles the specific API requirements such as SSE streaming and base64 encoding. The analysis found no security issues; the skill follows best practices for API key management and uses well-known service endpoints.
- Socketpass
No alerts
- Snykfail
Risk: HIGH · 1 issue
What does this agent skill do?
StepFun stepaudio-2.5-asr
Transcribe audio with StepFun's stepaudio-2.5-asr (released 2026-04, verified 2026-04-23). Long audio in one call, no chunking — but only if the request hits the right endpoint with the right body shape. The wrong endpoint returns an error that looks identical to "model doesn't exist", which is the #1 reason this skill exists.
Companion: for TTS with
stepaudio-2.5-tts(the sibling model), use thestepfun-ttsskill — they share an API key but live on different endpoints with different body shapes.
Why this skill exists — three traps that cost hours
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Wrong endpoint, wrong error.
stepaudio-2.5-asrdoes not live on/v1/audio/transcriptions(that endpoint serves the olderstep-asrfamily). It lives on/v1/audio/asr/sse— SSE streaming, JSON body, base64 audio. Sending it to the wrong endpoint returns{"error":{"message":"model stepaudio-2.5-asr not supported"}}, which is identical in structure to a genuinely nonexistent model name. People waste hours filing whitelist tickets. -
Plan key vs Normal key, silent failure. StepFun's "Plan" subscription keys (cheap, text-only) cannot call audio endpoints, but the failure manifests as a 4xx with no auth-shaped error message. If your account has a Plan subscription, you need a separate "Normal" key from the same console.
-
SSE error events are real. Censorship can fire on the ASR side too (rarely). Don't assume only
transcript.text.deltaandtranscript.text.doneevents arrive — handletype: errorevents in the stream or you'll silently drop them.
Config and auth
API key resolves in this order (fail-fast, no defaults):
$STEPFUN_API_KEYenvironment variable${CLAUDE_PLUGIN_DATA}/config.jsonwith{"api_key": "..."}(cross-session persistence)
First-time setup:
mkdir -p "${CLAUDE_PLUGIN_DATA}" && cat > "${CLAUDE_PLUGIN_DATA}/config.json" <<EOF
{"api_key": "<paste Normal key here>"}
EOF
If the user has not set a key, ask them to paste it — do not guess or use a placeholder. Get keys at https://platform.stepfun.com/ → API Keys. Use a Normal key, not a Plan key.
Quick start — single file
python3 scripts/asr_transcribe.py /path/to/audio.mp3
Output: plain text transcription on stdout.
For machine-readable output with usage / timing:
python3 scripts/asr_transcribe.py /path/to/audio.mp3 --json
For non-Chinese audio:
python3 scripts/asr_transcribe.py /path/to/audio.mp3 --language en
The script handles base64 encoding, the nested {audio: {data, input: {transcription, format}}} body, SSE parsing, and the misleading-endpoint pitfall. Prefer it over hand-rolled HTTP calls unless integrating into a larger pipeline.
Decision table
| Scenario | Action |
|---|---|
| Short clip (< 5 min), Chinese or English, mp3/wav/ogg/opus | python3 scripts/asr_transcribe.py audio.mp3 |
| Long audio (5-30 min) | Same script — 32K context handles it in a single call, no chunking needed |
| Audio > 30 min | Split with ffmpeg before sending; the API rejects oversized payloads |
| Need usage/billing data | Add --json to capture usage.input_tokens / usage.total_tokens from transcript.text.done |
| Highly repetitive content (same phrase 5+ times, > 90s) | Cross-validate with step-asr-1.1 — see repetition hallucination in references/known_issues.md |
Hit model stepaudio-2.5-asr not supported | Wrong endpoint. Switch from /v1/audio/transcriptions to /v1/audio/asr/sse |
| Hit silent 4xx auth failure | Verify your key is "Normal" not "Plan" — Plan keys cannot call audio endpoints |
| Need to write raw HTTP (no Python) | Read references/api_reference.md for exact JSON body and SSE event shapes |
Supported audio formats
The script auto-detects from extension; pass --format to override:
| Extension | Format flag | Notes |
|---|---|---|
.mp3 | mp3 | Most common, default |
.wav | wav | Lossless |
.ogg | ogg | OGG container |
.opus | ogg | Opus codec in OGG container — pass through unchanged |
.pcm | pcm | Raw PCM — also requires format.rate, format.channel, format.bits (see API reference) |
For mp4/m4a/webm/etc., transcode to one of the above first via ffmpeg. Production pipelines often pre-transcode everything to OGG/Opus 16kHz mono to minimize base64 payload size.
Capacity and performance (verified 2026-04-23)
- 32K context window — single-call upper limit, no chunking needed for ≤ 30 min audio
- ~85-101× RTF on long audio (17.4 min audio → 10.4s wall clock)
- ~5.3× speedup vs step-asr-1.1 at the 100s+ length range
- Only ~2× speedup at the 5-15s range — the LLM spin-up cost dominates short clips. If your workload is many short clips, the migration ROI is modest
Common error patterns
| Error response | Actual cause | Fix |
|---|---|---|
"model stepaudio-2.5-asr not supported" on /v1/audio/transcriptions | Wrong endpoint | Switch to /v1/audio/asr/sse (script does this) |
| Silent 4xx with no auth message | Using a "Plan" key on audio endpoint | Get a "Normal" key from the StepFun console |
| ASR returns 3-4× expected character count | Repetition hallucination on highly-repetitive audio | Cross-validate with step-asr-1.1; see references/known_issues.md |
data: {"type":"error","message":"content blocked..."} mid-stream | Censorship fired on user-uploaded content | Handle SSE error event explicitly; don't assume only delta/done arrive |
More edge cases in references/known_issues.md.
Design invariants (do not break)
- Always pass through SSE — don't try to buffer the response with a non-streaming client. The model emits
transcript.text.deltafor long audio;transcript.text.donecarries the authoritative full text andusage. Reject the SSE format entirely and you'll get nothing. - Take final text from
transcript.text.done.text— concatenated deltas can drift on edge cases. Deltas are for progressive UI; thedoneevent is the source of truth. - Handle
errorevents in the stream — don't treat the SSE stream as if only success events arrive. A blocked-content event mid-stream returnstype: errorwith nodoneevent. - Fail-fast on missing API key — never default to a placeholder or empty string. The script does this; preserve the behavior in any wrapper.
When to read references
references/api_reference.md— exact JSON request body, all fields, all SSE event types, response examples. Read when writing raw HTTP calls instead of using the bundled script.references/known_issues.md— repetition hallucination details, the wrong-endpoint diagnostic trail, Plan-vs-Normal key gotcha, ASR-side censorship handling, pricing opacity. Read when debugging anomalous output or evaluating whether to migrate fromstep-asr-1.1.
Pricing (verified 2026-04-23, volatile)
stepaudio-2.5-asr is in invitation beta as of 2026-04-23 — no public per-minute rate. The step-asr-1.1 baseline is 2.2 元/小时. The invitation PDF mentions "成本直降 80%" implying ~0.4 元/小时, but this is not yet on the pricing page. Re-verify at https://platform.stepfun.com/docs/zh/guides/pricing/details before quoting to stakeholders.
Next Step
ASR transcripts almost always benefit from one of two downstream steps. After producing the transcript, suggest the natural next skill:
Transcription complete. The output is raw text from the model — common next steps:
Options:
A) transcript-fixer — clean up ASR errors (homophones, segmentation, filler words). Recommended if the recording is a real-world conversation, podcast, or interview rather than read-aloud text
B) meeting-minutes-taker — turn the transcript into structured minutes with decisions, action items, and speaker attribution. Recommended if the recording is a meeting
C) No thanks — the raw transcript is what I needed
Skip the suggestion when the user has already specified the downstream tool, or when the transcription was clearly a one-off lookup (e.g., "what does this 15-second clip say?").
How can the creator link this skill?
Add the canonical catalog link to the repository README so users can inspect current installs and available audits. The publishing guide covers the complete discovery path.
<a href="https://skillzs.dev/skills/daymade/claude-code-skills/stepfun-asr">View stepfun-asr on skillZs</a>