scriba

Imported from AlexanderAbramovPav/scriba (SKILL.md). Install upstream with npx skills add AlexanderAbramovPav/scriba. Copyright stays with the author.

scriba

Turns any audio/video file into an accurate, speaker-diarized Markdown transcript. Two-pass: first produce Speaker 1/2/3 + representative samples, then rename to real people.

First-run setup — the AI handles it; the user only clicks 3 links and pastes a token

Before you call transcribe.sh for the first time on this machine, check whether ~/.config/scriba/hf_token exists. If it does, skip this block. If it doesn't, the user needs a HuggingFace token before diarization can work — and they should NOT be left to figure that out alone. Walk them through it conversationally in their language (the chat's language is the user's language; default to English when ambiguous). The flow:

«Чтобы определить, кто что говорит, нужен бесплатный HuggingFace-токен (~30 секунд):

1. Открой https://huggingface.co/join — заведи аккаунт, если нет.
2. Открой https://huggingface.co/pyannote/segmentation-3.0 и нажми «Agree and access repository» (один клик).
3. То же на https://huggingface.co/pyannote/speaker-diarization-community-1 — это модель, которая распознаёт спикеров (один клик).
4. Открой https://hf.co/settings/tokens → «+ Create new token» → имя scriba, тип Read, скопируй сгенерированный токен (вид hf_...).
5. Вставь его сюда в чат — я сохраню локально с правами 600 и больше никогда не попрошу.»

When the user pastes the token, write it via Bash:

mkdir -p ~/.config/scriba && umask 077 && printf '%s\n' '<TOKEN>' > ~/.config/scriba/hf_token && chmod 600 ~/.config/scriba/hf_token

Then continue with the transcription. Tell the user the token is read-only and can be revoked at any time from https://hf.co/settings/tokens.

Without the token transcription still works, you just get one collapsed Speaker 1 for everyone. If the user explicitly says they don't want to set up HF, proceed without the token — but warn them once that speakers won't be separated.

Prereqs check. Before transcribing, also verify bash, Python 3.10+, uv, and ffmpeg/ffprobe are on $PATH. If any are missing, name the missing tool to the user and give them the exact install command for their OS (brew install ffmpeg on macOS, curl -LsSf https://astral.sh/uv/install.sh | sh for uv on either OS, etc.). Don't try to install for them — just tell them what to run.

First invocation of transcribe.sh auto-bootstraps .venv and downloads model weights (~5 min, ~3 GB). Subsequent runs reuse the cache and start in seconds.

Modes

• Default = accuracy: whisperX large-v3 on CPU + pyannote community-1. Strong "who said what" — community-1's authors report ~10–11% DER on standard benchmarks, and the model is chosen to run on a normal Mac, not only a top-end one. Don't claim a scriba-specific accuracy number you haven't measured; the user can measure DER on their own labeled audio with scripts/benchmark_der.py (see benchmarks/). Slower — a ~1 h meeting ≈ 1–2 h of compute. Run long files in the background (e.g. via a background shell).
• --fast: MLX (GPU) transcription, coarser speaker boundaries. Use when speed matters.

Workflow

Pass 1 — transcribe

Run (background for long files; don't pipe to tail -N — it buffers until EOF and hides progress):

bash skills/scriba/scripts/transcribe.sh "<media-file>" [--fast] [--speakers N] [--lang XX]

This writes the transcript into a portable folder <title>.transcript/ next to the input (<title>.md + a data/ subfolder with the JSON sidecar and per-speaker voice clips) and prints the folder path. <title> is the input filename kebab-cased; when the filename is generic (zoom_0, GMT…, recording, a bare date), pass --title "<meaningful name>" and it's used instead. To decide, you can check the filename with python3 skills/scriba/scripts/naming.py generic "<input-stem>" (prints 1 if generic). It also prints a banner with the wall-clock ETA and the paths of two live-status files (see "Monitoring" below). Full layout: references/file-layout.md.

Glossary biasing (mixed RU/EN terminology)

Domain terms get mangled by ASR — product names, people, acronyms, and English tech words spoken inside Russian. Bias the model toward the right spellings: assemble these terms from the meeting's context (invite, agenda, prior transcripts in the same folder) and write them one per line to <recordings-dir>/.scriba/glossary.txt (next to the media). Blank lines and # comments are ignored. They feed initial_prompt/hotwords and bias every run in that folder. A global fallback list lives at ~/.config/scriba/glossary (project terms take precedence). Maintaining correct spellings here is the cheapest lever for mixed-terminology accuracy.

Known-speaker enrollment (skip "who is Speaker N?")

If you have a short single-speaker reference clip per known person, pass them so their speakers come out pre-named by construction instead of Speaker N:

bash skills/scriba/scripts/transcribe.sh "<media-file>" --enroll "Alice=alice.wav,Bob=bob.wav"

Matching uses community-1 voice embeddings + a cosine threshold: each meeting cluster is matched to its best reference above threshold (greedy, so a name claims at most one cluster). Unmatched speakers stay Speaker N and go through the normal identify/rename flow. For matched speakers, skip the "who is Speaker N?" question.

Persistent voiceprints vs per-recording clips. Reusable reference clips ("voiceprints") live in .scriba/voiceprints/ (project, next to the media) over ~/.config/scriba/voiceprints/ (global), and feed --enroll across meetings. These are distinct from the per-recording listening clips embedded at <title>.transcript/data/speaker-N.wav (those are just for this one meeting's ID). After the user names a new speaker, you may offer to save a reference clip as a voiceprint so that person is auto-recognised in future meetings.

Monitoring a running transcription

Default UX — point the user at the external surfaces, don't poll from the AI side. Status visibility is free this way (zero AI tokens) and works whether the user keeps the chat open or not:

• In the Claude Code chat's own statusline (preferred — same window, no extra terminal needed). The skill ships scripts/statusline.sh, which prints 🎙 <stage> <pct>%* · ETA MM:SS while a transcription runs (* = real, from a whisperX Transcript: [...] line; ~ = extrapolated, used in warmup / MLX path) and stays silent otherwise. To make it show up in the chat's bottom strip the user has to wire the script into their statusline once — recipes for claude-hud, the vanilla Claude Code statusLine, tmux, Starship, p10k, fish/zsh are in references/statusline-integration.md. The skill itself does not edit anyone's settings, so this is opt-in; mention it to the user the first time, point them at the file, and don't re-pitch. If they've already wired it but see nothing, the most common cause is they haven't restarted the Claude Code session since editing settings.json.
• For passive use (launch and walk away): macOS Notification Center pings the user automatically when stage=done (Glass sound, with the output path). Tell the user: «нотификация прилетит, когда готово».
• For active watching: bash <skill>/scripts/transcribe.sh --watch "<media-file>" — full-screen TUI in a side terminal: progress bar, audio %, ETA, last log line, refreshes every 2 s. Detach with Ctrl-C, transcription keeps running. Zero AI tokens.
• For a quick poke: tail -f "<input-stem>.transcript.log" — raw whisperX output streaming, including each Transcript: [start --> end] text segment as it lands.

Tell the user these options when you launch. Do NOT routinely poll from inside the chat — every Monitor event or --status call costs AI tokens, and over a long run the cache misses add up fast.

When AI-side polling is appropriate (the user explicitly asks "как там?", or you need to gate on completion before moving to the speakers step):

• One-shot: bash <skill>/scripts/transcribe.sh --status "<media-file>" → single line stage · elapsed · ETA (src) · audio % (src) · wall % · last_log. Cheap.
• Streaming, very long run: spawn ONE Monitor that emits only on stage changes and 25 % audio boundaries — not 10 %, not on every tick. That keeps event count ≤ 6–8 for a typical meeting.

Never use /loop, ScheduleWakeup, or any periodic wake-up to check on the transcription — every wake-up is a guaranteed prompt-cache miss (~10 K tokens) and adds up to dollars over a long run with no benefit you don't get from the silent macOS notification.

Schema for <stem>.transcript.progress.json (small, ~350 B, refreshed every 5 s)

During align and diarize whisperX is silent for tens of seconds. The wrapper emits → <step>… (Ns) heartbeats every 10 s to keep the log visibly alive — that's how you can tell pyannote hasn't hung. The statusline (scripts/statusline.sh) shows 🎙 align · 00:25 / 🎙 diarize · 00:35 with the in-stage timer for the same reason.

Full ETA model and chip table in references/eta-factors.md.

Note on Claude Code's "Shell details" view: that inspector shows a snapshot of the bash task — runtime and output don't auto-refresh. For live status, point the user at the statusline (3 s refresh) or --watch TUI (2 s refresh). Re-opening "Shell details" pulls a fresh snapshot.

Identify speakers

Open the transcript and read the ## Speakers — identify who's who section to the user in their chat language (translate the headings on the fly if the user chats in Russian/etc; the section is always written in English in the file). For each Speaker N show: the talk-time %, the path to the embedded data/speaker-N.wav clip (so they can listen if the text quote isn't enough), and the three textual samples. Ask who each speaker is and wait for their mapping (e.g. "Speaker 1 = Alice, Speaker 2 = Bob"). The audio clip is the most reliable ID signal — point at it explicitly for users who don't recognise the voice from text alone.

Pass 2 — rename

python3 skills/scriba/scripts/rename_speakers.py "<title>.transcript/<title>.md" "Иван,Аня"
# or explicit:
python3 skills/scriba/scripts/rename_speakers.py "<title>.transcript/<title>.md" --map "Speaker 1=Иван,Speaker 2=Аня"

Rules

• Always run the first-run setup check above before invoking transcribe.sh on a new machine. Don't skip it and don't fail silently to diarization-less mode — the user will be confused why everyone is "Speaker 1". Walk them through the HF-token onboarding the first time only; the file's presence is the cue to skip on later runs.
• Match the user's language in conversation (Russian / English / etc. — whatever they chat in). The transcript text itself stays verbatim in whatever language the audio was in.
• Never invent speaker names — always get them from the user via the samples (text quote and embedded <audio> clip).
• For long inputs, launch pass 1 in a background shell and tell the user it's running; don't block.
• After launching, tell the user about the external UI surfaces — the macOS notification (fires on done) and the --watch TUI command — so they can see progress without burning AI tokens. One short line is enough, in their language. Example (RU): «Запустил. Нотификация прилетит на готовности; для живой картинки — bash <skill>/scripts/transcribe.sh --watch "<file>" в соседнем терминале.»
• After that wait silently for the bg task to complete — do not periodically poll --status and do not spawn long-running Monitors unless the user explicitly asks for in-chat updates. The bg task notification fires when it's done; that's your cue to read the transcript.
• If the user explicitly asks for in-chat progress, prefer ONE Monitor over <stem>.transcript.progress.json with a 25 % audio threshold and stage-change emissions — not 10 %, not every tick. Stop the Monitor once you see stage=done.
• Never use /loop, ScheduleWakeup, or any timed wake-up to poll the transcription — every wake-up is a prompt-cache miss (~10 K tokens) and provides nothing you don't already get for free from the macOS notification.
• Do NOT read the bg-task stdout file or the .transcript.log for status — they're verbose. The cheap channels are --status (one line) and *.progress.json (small object).
• When presenting speakers for naming, show both the text samples AND the embedded <audio> clip path — the audio is the real ID signal for non-technical users.