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Integrate our audio recordings (mono or two-channel) with the Wavesift HTTP API. Build a small, well-tested client module (or CLI) in this project's primary language and follow the contract below exactly — do not invent endpoints or fields.
BASE URL: https://api.wavesift.com/v1
SPEC: the exact machine-readable contract (request/response schemas, enums, status codes) is the OpenAPI document at https://api.wavesift.com/docs/public/openapi.json (the human-readable contract is at /docs). Fetch it first and treat it as the source of truth where this text is less specific; do not use endpoints that are not in it.
AUTH: send header `X-Api-Key: <key>` on every request. Read the key from the env var WAVESIFT_API_KEY. Never log it, never commit it, never put it in a URL. Keys are created once by a human from a login session (POST /auth/login → Bearer token → POST /user/api-keys `{ "name", "expiresAt"? }` → `apiKey` is shown once); the integration itself must not create or rotate keys.
FORMAT: JSON with camelCase fields; enum values are lowercase snake_case strings; timestamps are ISO-8601 UTC. Errors are RFC 7807 `application/problem+json` with `status` and `detail`.
FLOW (one recording = one transcription):
1. Upload: POST /transcriptions as multipart/form-data with field `system_file` (the recording, mono or stereo; accepted extensions: wav, flac, mp3, m4a, m4b, aac, ogg, opus, wma, webm, mp4, mkv), optional `title`, `external_id` (our call id; the server dedupes on it — re-uploading with the same external_id returns the existing transcription instead of creating another one) and `language` (ISO code like "uk", or "auto"; default auto). Response 200 is the transcription object with `id`, `kind: "upload"` and `status: "queued"`. Body limit is 6 GB per request. Batch alternative: POST /transcriptions/bulk with up to 50 file fields → `{ items: [{ index, fileName, transcription | null, error | null }] }` (partial success is normal).
Channel handling: recordings can be mono or two-channel. When a two-channel file carries a different speaker on each channel, the server splits it automatically: LEFT channel → `speakerLabel: "SPEAKER_1"`, RIGHT channel → `"SPEAKER_2"`; these labels are fixed once the transcription is `completed`. For mono, or when both channels carry the same mix, there is no split: right after `completed` the segments have `speakerLabel: null`, then the server runs voice diarization and fills `SPEAKER_N` in order of first appearance; `audioDeletedAt` becoming non-null signals that this step is done.
2. Wait: poll GET /transcriptions/{id} every 5–10 s (or subscribe to GET /transcriptions/{id}/events, text/event-stream, events `status`/`progress`). `status` is one of queued | processing | completed | failed. On `failed` read `errorMessage` and re-upload; do not retry in a tight loop.
3. Read the transcript: GET /transcriptions/{id}/segments → `{ items: [{ id, start, end, text, source, speakerUserId, speakerLabel }] }` ordered by `start` (seconds). Or GET /transcriptions/{id}/markdown for text/markdown with timecodes and speaker tags like `[OTHER-SPEAKER_1]` (404 until completed).
4. Summary: POST /transcriptions/{id}/summaries with JSON `{ "presetId": "<guid>" }`. Only valid when the transcription is `completed` (otherwise 409). List presets with GET /summary-presets → `{ items: [{ id, name, prompt, isGlobal, ... }] }`; the ready-made two-speaker call preset is named "Розбір дзвінка КЦ" (id 0198c0de-0000-7000-8000-000000000005). To use our own instructions create a preset once: POST /summary-presets `{ "name", "prompt" }` (the server prepends a short preamble describing the transcript format and appends the transcript itself; do not include the transcript in the prompt; the summary is written in the transcript's language unless the prompt says otherwise) and reuse its id.
5. Wait for the summary: poll GET /summaries/{summaryId} every 5 s until `status: "completed"`, then read `markdown`. `failed` carries `errorMessage`.
WEBHOOKS (preferred over polling): register once with POST /user/webhooks `{ "url": "https://…", "events": ["transcription.completed", "transcription.diarized", "transcription.failed", "summary.completed", "summary.failed"] }`; the response contains `secret` (shown once, store it with the API key). Every delivery is a POST with JSON `{ id, event, occurredAt, data }` and headers `X-Wavesift-Event`, `X-Wavesift-Delivery`, `X-Wavesift-Timestamp`, `X-Wavesift-Signature: v1=<hex>` where hex = HMAC-SHA256(secret, "<timestamp>.<raw body>"). Verify the signature on the raw body with a constant-time compare, reject timestamps older than 5 minutes, respond 2xx within 10 s and process asynchronously, dedupe by delivery `id` (a delivery may be repeated; retries come after 1 min, 5 min, 30 min, 2 h, 12 h). For mono recordings speaker labels arrive with `transcription.diarized`, not with `transcription.completed`. Keep polling as a fallback only.
OTHER ENDPOINTS: GET /user/me (identity check), GET /transcriptions?page=&limit=&status=&kind=&q= (own history, `{ items, totalCount }`; `kind` is `upload` for single-file uploads and `meeting` when both mic_file and system_file were sent), GET /transcriptions/{id}/summaries (summary history), DELETE /transcriptions/{id} (204), GET /subscription/current (plan and usage; 0 in the plan and null in the remaining fields mean unlimited).
ERROR HANDLING: 400 → fix the request; 401 → the key is missing/invalid/expired, stop and alert a human (do not retry); 403 → wrong resource id or wrong auth method; 404 → unknown id or transcript not ready yet; 409 → business rule (transcription not completed, plan limit on file size / minutes / presets) — wait or surface it; 413 → file over 6 GB (the body may be an HTML page from the proxy); 429 → rate limit (10/min per IP on /auth/*, 300/min per account elsewhere), wait for the seconds in the `Retry-After` header and never poll more often than every 5 s; 5xx → retry with exponential backoff (max 5 attempts) and include the `X-Correlation-Id` response header in logs. Set the correlation header yourself (a UUID per call) so we can trace requests.
REQUIREMENTS: keep our original audio (the server deletes source audio after processing); store `transcriptionId`, `summaryId`, transcript markdown and summary markdown against our call record; send our call id as `external_id` so retries never create duplicates; unit-test the JSON mapping and the status machine with recorded fixtures; add a CLI/command that runs the whole flow for one file and prints the summary.