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daymade/claude-code-skills321 installs

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-asr
view source ↗

Is 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 the stepfun-tts skill — they share an API key but live on different endpoints with different body shapes.

Why this skill exists — three traps that cost hours

  1. Wrong endpoint, wrong error. stepaudio-2.5-asr does not live on /v1/audio/transcriptions (that endpoint serves the older step-asr family). 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.

  2. 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.

  3. SSE error events are real. Censorship can fire on the ASR side too (rarely). Don't assume only transcript.text.delta and transcript.text.done events arrive — handle type: error events in the stream or you'll silently drop them.

Config and auth

API key resolves in this order (fail-fast, no defaults):

  1. $STEPFUN_API_KEY environment variable
  2. ${CLAUDE_PLUGIN_DATA}/config.json with {"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

ScenarioAction
Short clip (< 5 min), Chinese or English, mp3/wav/ogg/opuspython3 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 minSplit with ffmpeg before sending; the API rejects oversized payloads
Need usage/billing dataAdd --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 supportedWrong endpoint. Switch from /v1/audio/transcriptions to /v1/audio/asr/sse
Hit silent 4xx auth failureVerify 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:

ExtensionFormat flagNotes
.mp3mp3Most common, default
.wavwavLossless
.oggoggOGG container
.opusoggOpus codec in OGG container — pass through unchanged
.pcmpcmRaw 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 responseActual causeFix
"model stepaudio-2.5-asr not supported" on /v1/audio/transcriptionsWrong endpointSwitch to /v1/audio/asr/sse (script does this)
Silent 4xx with no auth messageUsing a "Plan" key on audio endpointGet a "Normal" key from the StepFun console
ASR returns 3-4× expected character countRepetition hallucination on highly-repetitive audioCross-validate with step-asr-1.1; see references/known_issues.md
data: {"type":"error","message":"content blocked..."} mid-streamCensorship fired on user-uploaded contentHandle SSE error event explicitly; don't assume only delta/done arrive

More edge cases in references/known_issues.md.

Design invariants (do not break)

  1. Always pass through SSE — don't try to buffer the response with a non-streaming client. The model emits transcript.text.delta for long audio; transcript.text.done carries the authoritative full text and usage. Reject the SSE format entirely and you'll get nothing.
  2. Take final text from transcript.text.done.text — concatenated deltas can drift on edge cases. Deltas are for progressive UI; the done event is the source of truth.
  3. Handle error events in the stream — don't treat the SSE stream as if only success events arrive. A blocked-content event mid-stream returns type: error with no done event.
  4. 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 from step-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?").

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>