version: "1.0.0" name: audiocpp-timed-transcript description: "Industry-standard time-coded transcripts (SRT/WebVTT/JSON) via GPU host AudioCPP: product ASR for text + private qwen3_forced_aligner (/v1/tasks/run) for word times, collapsed to caption cues. Use when the user wants timed captions without PyTorch." metadata: guideants: enabled: true display_order: 35 requires_toolsets: [sandbox]

audio.cpp time-coded transcripts (experimental)

Paths — fixed layout, do not probe or re-derive. The sandbox CWD is the notebook's output directory. This skill's scripts live under Skills/audiocpp-timed-transcript/scripts/ relative to it. Write deliverables with bare filenames (e.g. -o meeting); never prefix with Output/.

Product ASR returns text only. Word timings come from AudioCPP’s qwen3_forced_aligner over the GPU host raw gateway (/private/v1/tasks/run), then words are grouped into SRT / WebVTT / segment JSON (industry caption formats).

Authority: upstream audio.cpp model_specs/qwen3_forced_aligner.json, docs/models/qwen3.md, app/server/runtime.cpp (/v1/audio/transcriptions is text-only; timed fields are on /v1/tasks/run).

Environment (PC → the GPU host)

AUDIOCPP_SKILL_BASE_URL=http://<gpu-host-lan-ip>:8112/audiocpp-skill
AUDIOCPP_SKILL_TOKEN=<same as the GPU host GA_AUDIOCPP_SKILL_TOKEN>

ASR must already be loaded on the GPU host (Settings / API lifecycle).

Phase 0 gate (once per GPU host image)

python3 Skills/audiocpp-timed-transcript/scripts/fetch_model.py \
  audio-cpp/audio.cpp-gguf \
  --dest /models-local/skill/asr/Qwen3-ForcedAligner-0.6B-GGUF \
  --include Qwen3-ForcedAligner-0.6B-GGUF/qwen3-forced-aligner-0.6b-q8_0.gguf \
  --strip-prefix Qwen3-ForcedAligner-0.6B-GGUF/
 
# ROCm host builds need --backend rocm (not cuda)
python3 Skills/audiocpp-timed-transcript/scripts/spawn_engine.py start \
  --path /models-local/skill/asr/Qwen3-ForcedAligner-0.6B-GGUF \
  --family qwen3_forced_aligner --task align --backend rocm
 
python3 Skills/audiocpp-timed-transcript/scripts/spawn_engine.py status

If status is not ready (e.g. missing model spec), update the GPU host guideants-ai audiocpp image. Do not add a PyTorch aligner.

Transcribe

python3 Skills/audiocpp-timed-transcript/scripts/timed_transcribe.py \
  uploads/clip.wav -o clip --language English

Writes clip.srt, clip.vtt, clip.json, clip.transcript.txt.

When finished: spawn_engine.py stop.

Limits

• Standalone ForcedAligner does not chunk internally; this script chunks at

30 s by default (matches Qwen3 ASR audio_chunk_seconds). Cap --max-chunk-s at 60 — longer passes hit max_source_positions.

• Long-form: raise --budget-seconds (e.g. 900) for multi-chunk runs.
• Aligner languages: 11 (pass --language explicitly).
• Cue text is grouped from word times (~3–8 s); not Whisper-native segment IDs.
• ROCm hosts: spawn_engine.py start ... --backend rocm.
• Multi-model private engine (align+diar): spawn_engine.py start ... --extra path=...;family=...;task=...

requires the updated skill gateway on the GPU host (Gate M).

Related

Speaker-labeled timed captions: audiocpp-diarize (diarize.py + merge_diarized.py).

Reporting

End by quoting spawn/status evidence and which output files were written.