stepfun-asr
GitHub调用阶跃音频ASR模型进行语音转文字,支持中英文长音频。解决错误端点、API Key类型及SSE错误处理等陷阱。
Trigger Scenarios
Install
npx skills add daymade/claude-code-skills --skill stepfun-asr -g -y
SKILL.md
Frontmatter
{
"name": "stepfun-asr",
"description": "Transcribes Chinese\/English audio with StepFun's stepaudio-3-asr-max via its SSE endpoint (not \/v1\/audio\/transcriptions) — one call handles long-form audio with no chunking. Use when migrating from step-asr\/stepaudio-2.5-asr, or hitting the misleading \"model not supported\" error (actually wrong endpoint). Triggers on 阶跃 ASR, 语音识别. Not for TTS with the sibling model (use stepfun-tts).",
"disable-model-invocation": true
}
StepFun stepaudio-3-asr-max
Transcribe audio with StepFun's stepaudio-3-asr-max (StepAudio 3, released 2026-09-15, verified 2026-09-16; supersedes stepaudio-2.5-asr on the same endpoint). 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-3-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
-
Wrong endpoint, wrong error.
stepaudio-3-asr-maxdoes 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-3-asr-max 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
Per-word timestamps need no flag — --json always carries segments:
"segments": [{"text": "Understand ", "start_ms": 228, "end_ms": 1108}, ...]
One entry per word, monotonic. A few words share their predecessor's timestamp (the server flushes in blocks), which is fine for locating a moment but not for forced alignment.
To use an older model:
python3 scripts/asr_transcribe.py /path/to/audio.mp3 --model stepaudio-2.5-asr
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 |
| Need to know when each word was said | --json, read segments. On by default |
| Need speaker labels (who said what) | python3 scripts/asr_file.py <public-url> — a different, async endpoint. Takes a URL, not a local file: base64 and StepFun's own file store are both rejected, so hosting the audio somewhere fetchable is a decision for whoever runs it |
--model stepaudio-2-asr-pro returns internal error |
That model is not usable on /v1/audio/asr/sse (measured 2026-09-18); use the default or stepaudio-2.5-asr |
| Highly repetitive content (same phrase 5+ times, > 90s) | Cross-validate with step-asr-1.1 — see repetition hallucination in references/known_issues.md (2.5-era issue, unverified on v3) |
Hit model stepaudio-3-asr-max 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 |
Speaker labels — scripts/asr_file.py
stepaudio-3-asr-max on /v1/audio/asr/sse has no speaker capability at all (14 candidate
request fields measured inert). Diarization lives on the async file endpoint:
python3 scripts/asr_file.py https://example.com/talk.mp3
# [ 6.61- 8.43] speaker_0: Hello. Hello. Oh,
# [ 8.21- 10.11] speaker_1: hello! I didn't know you were there.
Verified end-to-end 2026-09-18 on a two-speaker sample: correct turn boundaries, per-word
timestamps inside each utterance, up to 10 speakers per task. Uses stepaudio-2.5-asr —
v3 is not served on this endpoint.
The hard constraint: it fetches a URL and nothing else. Base64 is rejected and so is
StepFun's own stepfile:// file store, so there is no way to feed it a local file without
first putting that file somewhere publicly fetchable. Treat that as the caller's decision.
references/known_issues.md has the three dead ends and the retry/redirect behaviour.
Parameters are free — never omit one silently
Sending more request parameters costs nothing: billing is per audio-hour. So the default is
send everything useful, and every field we do not send has to carry a written reason in
REQUEST_PARAMS at the top of scripts/asr_transcribe.py.
python3 scripts/check_params.py # diff official field table vs REQUEST_PARAMS
python3 scripts/check_params.py --selftest # calibrate the check before trusting it
Two guards, one per direction:
- Request side —
check_params.pyfetches the official field table and fails if it lists a fieldREQUEST_PARAMSdoes not mention.--selftestcalibrates both ways: the real manifest must pass (no false alarms), and a manifest withenable_timestampremoved must be caught (the actual historical gap — per-word timestamps were missing for months because only the response field table was ever read). - Response side — the parser reports
unhandled_response_fieldsfor anything the server sends that it does not consume, because that is what the timestamp gap looked like from this side:start_time/end_timearrived on every delta and were thrown away.
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 pass --rate, --bits, --channel (and --codec) |
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
v3 spot measurements (verified 2026-09-16): 10s clip → 1.1s, 53s real-world clip → 2.6s (~20× RTF). v2.5-era baseline for reference (2026-04-23, same endpoint): 32K context window, ~85-101× RTF on 17.4 min audio, single-call ceiling ≈ 30 min — treat 30 min as the working ceiling for v3 until re-probed, and re-measure before quoting long-audio numbers.
Common error patterns
| Error response | Actual cause | Fix |
|---|---|---|
"model stepaudio-3-asr-max 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.transcribe()'s return dict always carries anerrorslist of the raw error-event payloads (empty if none fired), even whenokisTrue, so a caller wanting the old strict semantics (any error event ⇒ failure) must checkerrorsitself. - 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 2026-09-16, volatile)
stepaudio-3-asr-max: 2.8 元/小时 (official model page, 2026-09-16). The older stepaudio-2.5-asr remains served on the same endpoint as a fallback. 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?").
Version History
- c6903aa Current 2026-09-28 12:33
-
f7c2028
2026-09-23 01:32
修复 transcribe() 方法中 error 事件处理逻辑,统一返回 errors 字段以保留原始错误载荷,避免上游调用方因严格校验而误判失败;同步更新文档契约说明。
- e00a2ec 2026-08-20 11:25


