fegger
c4b86213d4
- final_transcribe gains optional speaker labeling and library upload: diarize_url runs a second tinydiarize pass and merges the turn times onto the quality transcript as 'Sprecher 1/2:' labels (diarize_merge, port of the Android Diarization object; failures keep the transcript unlabeled); library_url uploads the WAV + txt/srt/json bundle with the agenda — the server then generates summary + agenda coverage - WhisperServerEngine supports diarize mode (tinydiarize form field + speaker_turn_next parsing); multipart builder handles multiple files - CLI: --agenda-file, --diarize-url, --library-url (env vars MEETREC_DIARIZE_URL / MEETREC_LIBRARY_URL) - GUI: Library URL + Diarize URL fields and an agenda editor; upload progress lands in the status bar - validated end-to-end against the live servers: recording -> large-v3 transcription -> txt/srt/json -> library upload (all four files, device=desktop) -> server summary/agenda pipeline ran (honest error for a noise-only test recording)
823 lines
33 KiB
Python
823 lines
33 KiB
Python
#!/usr/bin/env python3
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"""
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meetrec.py — record an in-person meeting and transcribe it with Whisper.
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Records from any PipeWire/ALSA input (microphone, USB conference mic,
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USB audio interface, or a Pulse/PipeWire sink "monitor") while optionally
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printing a live rolling transcript. On Ctrl+C it stops cleanly and runs a
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final, higher-quality full transcription.
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Outputs (with -o meeting):
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meeting.wav raw recording, mono WAV at the input's native rate
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meeting.live.srt live rolling transcript (only when --live > 0)
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meeting.txt final transcript
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meeting.srt final transcript with timestamps
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meeting.json final transcript, structured
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Audio is recorded at the input device's native sample rate; both the live
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window and the engines get it resampled to Whisper's 16 kHz internally.
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Setup on Arch:
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sudo pacman -S python pipewire pipewire-pulse
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python -m venv .venv && source .venv/bin/activate
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pip install numpy sounddevice faster-whisper
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Usage:
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python meetrec.py --list-sources
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python meetrec.py -s "Conference Mic" --model small --live 8 -o meeting
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python meetrec.py -s monitor --model base --language en # record system audio
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python meetrec.py --engine whisper-cpp -o meeting # GPU via Vulkan (AMD/NVIDIA)
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Ctrl+C to stop -> final transcripts are written.
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"""
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import argparse
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import json
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import os
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import queue
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import shutil
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import subprocess
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import sys
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import tempfile
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import threading
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import time
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import urllib.error
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import urllib.request
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import wave
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from pathlib import Path
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from types import SimpleNamespace
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import numpy as np
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import sounddevice as sd
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SR = 16000 # sample rate — what Whisper wants
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LIVE_WINDOW = 60 # seconds of rolling audio kept for live transcription
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# --------------------------------------------------------------------------
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# Helpers
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# --------------------------------------------------------------------------
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def fmt_hms(t: float) -> str:
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h, rem = divmod(int(t), 3600)
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m, s = divmod(rem, 60)
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return f"{h:02d}:{m:02d}:{s:02d}"
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def fmt_srt(t: float) -> str:
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ms = int(round((t - int(t)) * 1000))
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h, rem = divmod(int(t) + (1 if ms >= 1000 else 0), 3600)
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m, s = divmod(rem, 60)
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if ms >= 1000:
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ms -= 1000
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return f"{h:02d}:{m:02d}:{s:02d},{ms:03d}"
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def write_srt(path: str, items) -> None:
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with open(path, "w") as f:
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for i, (a, b, text) in enumerate(items, 1):
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f.write(f"{i}\n{fmt_srt(a)} --> {fmt_srt(b)}\n{text}\n\n")
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def resample_16k(audio: np.ndarray, rate: int) -> np.ndarray:
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"""Linearly resample mono float32 audio to 16 kHz (Whisper's rate).
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Only needed for the live rolling buffer, which is fed to the engines as
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a raw array (they assume 16 kHz there). WAV files are resampled by the
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engines themselves.
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"""
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audio = np.asarray(audio, dtype=np.float32)
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if rate == SR or audio.size == 0:
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return audio
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n_out = max(1, int(round(audio.size * SR / rate)))
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t_out = np.arange(n_out, dtype=np.float64) * (rate / SR)
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return np.interp(t_out, np.arange(audio.size, dtype=np.float64),
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audio).astype(np.float32)
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def list_sources() -> None:
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print("PortAudio (PipeWire/Pulse/ALSA) input devices:\n")
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for i, d in enumerate(sd.query_devices()):
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if d["max_input_channels"] > 0:
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print(f" {i:>3} {d['name']}\n"
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f" api={d['hostapi']} in={d['max_input_channels']}ch "
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f"rate={d['default_samplerate']:.0f}")
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pactl = shutil.which("pactl")
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if pactl:
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print("\nPulseAudio/PipeWire sources (pactl):\n")
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try:
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out = subprocess.run(
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[pactl, "list", "short", "sources"],
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capture_output=True, text=True, timeout=5).stdout.strip()
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for line in out.splitlines():
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print(" " + line)
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except Exception:
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pass
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print("\nTip: sink monitors (record what the computer plays) appear as "
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"\"<sink name>.monitor\". Match them with -s <part of name>.")
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def resolve_source(name: str):
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"""Return a sounddevice device index, or None for the default device."""
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if name in ("default", None, ""):
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return None
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inputs = [(i, d) for i, d in enumerate(sd.query_devices())
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if d["max_input_channels"] > 0]
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matches = [(i, d) for i, d in inputs if name.lower() in d["name"].lower()]
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if not matches:
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print(f"error: no input device matching {name!r}. "
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f"Run with --list-sources.", file=sys.stderr)
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sys.exit(1)
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if len(matches) > 1:
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print(f"note: multiple devices match {name!r}, using: "
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f"{matches[0][1]['name']}", file=sys.stderr)
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idx, dev = matches[0]
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print(f"recording from: [{idx}] {dev['name']} ({dev['hostapi']})")
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return idx
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# --------------------------------------------------------------------------
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# Recording
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# --------------------------------------------------------------------------
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class Recorder:
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"""Records mono float32 at the input's native rate to a WAV file and
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keeps a rolling buffer for live transcription."""
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def __init__(self, device, wav_path: str):
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self.device = device
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self.wav_path = wav_path
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dev = sd.query_devices(kind="input") if device is None \
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else sd.query_devices(device)
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self.rate = int(dev["default_samplerate"]) # input's native rate
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self.q: "queue.Queue" = queue.Queue()
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self.lock = threading.Lock()
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self.buf = np.zeros(0, dtype=np.float32) # rolling window
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self.base = 0 # samples recorded before buf
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self.total = 0 # total samples recorded
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self.peak = 0.0 # peak amplitude of last block
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self.stop_event = threading.Event()
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self.stream = None
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self._writer = threading.Thread(target=self._write, daemon=True)
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self._writer.start()
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def _callback(self, indata, frames, time_info, status):
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if status:
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print(f"[rec] status: {status}", file=sys.stderr)
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self.peak = float(np.max(np.abs(indata[:, 0]))) if frames else 0.0
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self.q.put(indata.copy())
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with self.lock:
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block = indata[:, 0].astype(np.float32, copy=False)
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self.buf = np.append(self.buf, block)
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self.total += len(block)
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limit = LIVE_WINDOW * self.rate
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if len(self.buf) > limit:
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drop = len(self.buf) - limit
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self.buf = self.buf[drop:]
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self.base += drop
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def duration(self) -> float:
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with self.lock:
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return self.total / self.rate
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def _write(self):
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with wave.open(self.wav_path, "wb") as w:
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w.setnchannels(1)
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w.setsampwidth(2)
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w.setframerate(self.rate)
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while True:
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block = self.q.get()
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if block is None:
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break
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pcm = (np.clip(block[:, 0], -1.0, 1.0) * 32767).astype(np.int16)
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w.writeframes(pcm.tobytes())
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def snapshot(self):
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"""Return (rolling window resampled to 16 kHz, its start time in s)."""
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with self.lock:
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return resample_16k(self.buf, self.rate), self.base / self.rate
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def start(self):
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self.stream = sd.InputStream(
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samplerate=self.rate, channels=1, dtype="float32",
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device=self.device, callback=self._callback)
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self.stream.start()
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def stop(self):
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if self.stop_event.is_set():
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return # idempotent: the GUI close path may call this twice
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self.stop_event.set()
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self.q.put(None)
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if self.stream:
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self.stream.stop()
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self.stream.close()
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# The writer must drain the queue and close the WAV before we return,
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# otherwise the file can be truncated. Join in slices instead of one
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# long timeout so a slow disk still gets the full flush; only warn
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# after a total of 60 s (effectively never in practice).
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waited = 0.0
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while self._writer.is_alive() and waited < 60:
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self._writer.join(timeout=2.0)
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waited += 2.0
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if self._writer.is_alive():
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print("warning: WAV writer still running after 60 s; "
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"the file may be truncated", file=sys.stderr)
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# --------------------------------------------------------------------------
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# Transcription
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# --------------------------------------------------------------------------
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def live_worker(model, rec: Recorder, args, live_srt_path: str,
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stop: threading.Event, on_line=None):
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"""Transcribes the rolling window every `args.live` seconds and appends
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newly-finalized segments to the live .srt. If `on_line` is given, new
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lines are delivered to it (a, text) instead of being printed."""
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def emit(a, text):
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if on_line is not None:
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on_line(a, text)
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else:
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print(f"[{fmt_hms(a)}] {text}", flush=True)
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items = [] # (abs_start, abs_end, text) for the live .srt
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last_end = 0.0
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if on_line is None:
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print("live transcription on (Ctrl+C to stop)", flush=True)
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while not stop.is_set():
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stop.wait(args.live)
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if stop.is_set():
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break
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buf, base_s = rec.snapshot() # 16 kHz audio, absolute start time
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if len(buf) < 2 * SR: # need at least 2 s of audio
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continue
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try:
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segs, _info = model.transcribe(
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buf, beam_size=1, vad_filter=True, language=args.language)
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except Exception as e:
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print(f"[live] transcription error: {e}", file=sys.stderr)
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continue
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added = False
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for s in segs:
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a, b = base_s + s.start, base_s + s.end
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text = s.text.strip()
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if not text or b < last_end:
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continue
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items.append((a, b, text))
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last_end = max(last_end, b)
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emit(a, text)
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added = True
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if added:
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write_srt(live_srt_path, items)
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def final_transcribe(model, wav_path: str, out_stem: str, language,
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on_info=None, on_segment=None, on_done=None,
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diarize_url=None, agenda=None, library_url=None,
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device_name="desktop"):
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"""Full-pass transcription of the whole recording, with optional
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speaker labeling (tinydiarize server) and library upload.
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on_info(msg), on_segment(start, end, text) and on_done() are optional
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callbacks; when absent the same progress is printed to stdout.
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Returns (items, info): items are (start, end, text) tuples of the
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final — possibly speaker-labeled — transcript."""
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def info(msg):
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if on_info is not None:
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on_info(msg)
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else:
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print(msg, flush=True)
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def emit(a, b, text):
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if on_segment is not None:
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on_segment(a, b, text)
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else:
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print(f"[{fmt_hms(a)}] {text}", flush=True)
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info("\nRunning final transcription (this can take a while)...")
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segments, info_ = model.transcribe(
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wav_path, beam_size=5, vad_filter=True, language=language,
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vad_parameters=dict(min_silence_duration_ms=500))
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info(f"language: {info_.language} (p={info_.language_probability:.2f}) "
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f"duration: {fmt_hms(info_.duration)}")
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items = []
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if diarize_url:
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for s in segments: # collect first: labels need the whole pass
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items.append((s.start, s.end, s.text.strip()))
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info("Running speaker-diarization pass (tinydiarize server)...")
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try:
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tdrz = WhisperServerEngine(diarize_url, diarize=True)
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turns = tdrz.transcribe(wav_path, language=language)
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labeled = diarize_merge(
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[SimpleNamespace(start=a, end=b, text=t) for a, b, t in items],
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turns)
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items = [(s.start, s.end, s.text) for s in labeled]
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except Exception as e:
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info(f"diarize pass failed (continuing unlabeled): {e}")
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for a, b, text in items:
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emit(a, b, text)
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else:
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for s in segments: # generator: streams as it decodes
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text = s.text.strip()
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items.append((s.start, s.end, text))
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emit(s.start, s.end, text)
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full = " ".join(t for _, _, t in items)
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with open(out_stem + ".txt", "w") as f:
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f.write(full + "\n")
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write_srt(out_stem + ".srt", items)
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with open(out_stem + ".json", "w") as f:
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json.dump({
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"language": info_.language,
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"duration": round(info_.duration, 2),
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"segments": [{"start": round(a, 2), "end": round(b, 2), "text": t}
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for a, b, t in items],
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}, f, ensure_ascii=False, indent=2)
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info(f"\nwrote: {out_stem}.wav {out_stem}.txt {out_stem}.srt {out_stem}.json")
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if library_url:
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info("Uploading recording to the library...")
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try:
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rid = upload_recording(
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library_url, wav_path, out_stem, info_.duration,
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getattr(info_, "language", None), device_name, agenda or [])
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info(f"Uploaded to library as {rid} — summary and agenda check "
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"follow automatically on the server.")
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except Exception as e:
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info(f"upload failed: {e}")
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if on_done is not None:
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on_done()
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return items, info_
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# --------------------------------------------------------------------------
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# Engines
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# --------------------------------------------------------------------------
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#
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# Two interchangeable transcription backends:
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# faster-whisper CTranslate2 — NVIDIA CUDA or CPU (default)
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# whisper-cpp whisper.cpp CLI — GPU via Vulkan (AMD, Intel, NVIDIA)
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# or CPU; preferred on AMD hardware (e.g. Ryzen AI laptops)
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# whisper-server remote whisper.cpp HTTP server (server/whisper-server) —
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# the model lives on the server; ideal for phones/laptops
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#
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# Both expose: transcribe(audio, beam_size, vad_filter, language,
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# vad_parameters) -> (segments, info)
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# where `audio` is a 1-D float32 @16 kHz array or a WAV path, segments have
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# .start/.end/.text and info has .language/.language_probability/.duration.
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WHISPER_CPP_MODEL_FILES = {
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"tiny": "ggml-tiny.bin",
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"base": "ggml-base.bin",
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"small": "ggml-small.bin",
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"medium": "ggml-medium.bin",
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"large-v3": "ggml-large-v3.bin",
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}
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WHISPER_CPP_MODEL_URL = ("https://huggingface.co/ggerganov/whisper.cpp/"
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"resolve/main/{name}")
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WHISPER_CPP_VAD_FILE = "ggml-silero-v6.2.0.bin"
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WHISPER_CPP_VAD_URL = ("https://huggingface.co/ggml-org/whisper-vad/"
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"resolve/main/" + WHISPER_CPP_VAD_FILE)
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|
|
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def _download_file(url: str, dest: Path) -> None:
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tmp = dest.with_suffix(dest.suffix + ".part")
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req = urllib.request.Request(url, headers={"User-Agent": "meetrec"})
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with urllib.request.urlopen(req, timeout=60) as r:
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total = int(r.headers.get("Content-Length") or 0)
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done = 0
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with open(tmp, "wb") as f:
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|
while True:
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chunk = r.read(1024 * 1024)
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if not chunk:
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break
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f.write(chunk)
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done += len(chunk)
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if total:
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print(f"\r {done/1e6:.0f}/{total/1e6:.0f} MB", end="",
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flush=True)
|
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if total:
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print(flush=True)
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os.replace(tmp, dest)
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|
|
|
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def find_whisper_cpp_bin() -> str:
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"""Locate the whisper.cpp 'whisper-cli' binary."""
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|
candidates = [
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os.environ.get("WHISPER_CPP_BIN"),
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shutil.which("whisper-cli"),
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str(Path.home() / ".local/share/meetrec/whisper-cpp/bin/whisper-cli"),
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]
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|
for c in candidates:
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if c and os.path.isfile(c) and os.access(c, os.X_OK):
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return c
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|
raise RuntimeError(
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"whisper-cli not found. Build whisper.cpp (e.g. './install.sh "
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|
"--whisper-cpp', or cmake -DWHISPER_VULKAN=ON) or set WHISPER_CPP_BIN "
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|
"to the binary.")
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|
|
|
|
|
def _whisper_cpp_cache() -> Path:
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|
cache = Path(os.environ.get(
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|
"MEETREC_CACHE", str(Path.home() / ".cache/meetrec")))
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|
d = cache / "whisper-cpp"
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d.mkdir(parents=True, exist_ok=True)
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return d
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|
|
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def _cached_ggml(fname: str, url: str) -> str:
|
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"""Return the path of a cached GGML file, downloading it if missing."""
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p = _whisper_cpp_cache() / fname
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if p.exists() and p.stat().st_size > 0:
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return str(p)
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print(f"downloading {fname} to {p} ...", flush=True)
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_download_file(url, p)
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return str(p)
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|
|
|
|
def whisper_cpp_model_path(model_name: str) -> str:
|
|
"""Return the local path of a whisper.cpp GGML model, downloading it."""
|
|
try:
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fname = WHISPER_CPP_MODEL_FILES[model_name]
|
|
except KeyError:
|
|
raise RuntimeError(
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f"unknown model {model_name!r}; use one of: "
|
|
f"{', '.join(WHISPER_CPP_MODEL_FILES)}")
|
|
return _cached_ggml(fname, WHISPER_CPP_MODEL_URL.format(name=fname))
|
|
|
|
|
|
def write_wav_float(path: str, audio: np.ndarray) -> None:
|
|
"""Write a 1-D float32 mono @16 kHz array as 16-bit PCM WAV."""
|
|
audio = np.asarray(audio, dtype=np.float32).ravel()
|
|
pcm = (np.clip(audio, -1.0, 1.0) * 32767).astype(np.int16)
|
|
with wave.open(path, "wb") as w:
|
|
w.setnchannels(1)
|
|
w.setsampwidth(2)
|
|
w.setframerate(SR)
|
|
w.writeframes(pcm.tobytes())
|
|
|
|
|
|
def _parse_ts(s: str) -> float:
|
|
"""whisper.cpp timestamp 'HH:MM:SS,mmm' -> seconds."""
|
|
h, m, rest = s.split(":")
|
|
sec, ms = rest.split(",")
|
|
return int(h) * 3600 + int(m) * 60 + int(sec) + int(ms) / 1000.0
|
|
|
|
|
|
class WhisperCppEngine:
|
|
"""Transcription via the whisper.cpp 'whisper-cli' binary.
|
|
|
|
Build whisper.cpp with -DWHISPER_VULKAN=ON to run on a GPU (Radeon,
|
|
Intel, or NVIDIA); it falls back to CPU otherwise. GPU vs. CPU is decided
|
|
by the build, so --device/--compute-type do not apply. vad_filter=True
|
|
enables whisper.cpp's Silero VAD (auto-downloaded); vad_parameters are
|
|
accepted but not mapped.
|
|
"""
|
|
|
|
def __init__(self, model_name: str, cli: str = None):
|
|
self.model_name = model_name
|
|
self.cli = cli or find_whisper_cpp_bin()
|
|
self.model_path = whisper_cpp_model_path(model_name)
|
|
|
|
def transcribe(self, audio, beam_size=5, vad_filter=False,
|
|
language=None, vad_parameters=None):
|
|
# vad_parameters accepted for interface compatibility; Silero VAD uses
|
|
# whisper.cpp's own defaults
|
|
with tempfile.TemporaryDirectory(prefix="meetrec-") as td:
|
|
if isinstance(audio, (str, os.PathLike)):
|
|
wav = str(audio)
|
|
with wave.open(wav, "rb") as w:
|
|
duration = w.getnframes() / w.getframerate()
|
|
else:
|
|
wav = os.path.join(td, "in.wav")
|
|
write_wav_float(wav, audio)
|
|
duration = len(audio) / SR
|
|
|
|
cmd = [self.cli, "-m", self.model_path, "-f", wav,
|
|
"-oj", "-of", os.path.join(td, "out"),
|
|
"-bs", str(max(1, int(beam_size))), "-np",
|
|
"-l", language or "auto"]
|
|
if vad_filter:
|
|
cmd += ["--vad", "-vm", _cached_ggml(
|
|
WHISPER_CPP_VAD_FILE, WHISPER_CPP_VAD_URL)]
|
|
proc = subprocess.run(cmd, capture_output=True, text=True)
|
|
if proc.returncode != 0:
|
|
raise RuntimeError(
|
|
(proc.stderr or proc.stdout).strip()
|
|
or f"whisper-cli exited {proc.returncode}")
|
|
with open(os.path.join(td, "out.json")) as f:
|
|
data = json.load(f)
|
|
|
|
result = data.get("result", {})
|
|
segs = []
|
|
for item in data.get("transcription", []):
|
|
off = item.get("offsets", {})
|
|
if "from" in off and "to" in off: # milliseconds (current)
|
|
a, b = off["from"] / 1000.0, off["to"] / 1000.0
|
|
else: # 'HH:MM:SS,mmm' (older)
|
|
ts = item.get("timestamps", {})
|
|
a = _parse_ts(ts.get("from", "00:00:00,000"))
|
|
b = _parse_ts(ts.get("to", "00:00:00,000"))
|
|
text = item.get("text", "").strip()
|
|
if text:
|
|
segs.append(SimpleNamespace(start=a, end=b, text=text))
|
|
info = SimpleNamespace(
|
|
language=result.get("language") or language or "unknown",
|
|
language_probability=1.0, # not reported by whisper.cpp
|
|
duration=duration,
|
|
)
|
|
return segs, info
|
|
|
|
|
|
MULTIPART_BOUNDARY = "meetrec-9f3e1c7a5b2d"
|
|
|
|
|
|
def _multipart_body(fields: dict, files) -> bytes:
|
|
"""Build a multipart/form-data body with stdlib only.
|
|
files: list of (field_name, filename, file_obj) tuples."""
|
|
parts = []
|
|
for name, value in fields.items():
|
|
parts.append(
|
|
f"--{MULTIPART_BOUNDARY}\r\n"
|
|
f'Content-Disposition: form-data; name="{name}"\r\n\r\n'
|
|
f"{value}\r\n".encode())
|
|
for name, fname, fh in files:
|
|
parts.append(
|
|
f"--{MULTIPART_BOUNDARY}\r\n"
|
|
f'Content-Disposition: form-data; name="{name}"; '
|
|
f'filename="{fname}"\r\n'
|
|
f"Content-Type: application/octet-stream\r\n\r\n".encode())
|
|
parts.append(fh.read())
|
|
parts.append(b"\r\n")
|
|
parts.append(f"--{MULTIPART_BOUNDARY}--\r\n".encode())
|
|
return b"".join(parts)
|
|
|
|
|
|
class WhisperServerEngine:
|
|
"""Transcription via a remote whisper.cpp server (POST /inference).
|
|
|
|
Use the Docker Compose stack in server/whisper-server/ to run one
|
|
(Vulkan GPU on AMD, or CPU). The model lives on the server, so
|
|
--model / --device / --compute-type do not apply, and beam size is a
|
|
server-start setting in whisper.cpp v1.9.3 (no per-request override) —
|
|
beam_size is accepted for interface compatibility but ignored.
|
|
vad_filter/vad_parameters are likewise accepted but unused; the server
|
|
runs its own pipeline. Note: verbose_json reports language names like
|
|
"german" rather than ISO codes.
|
|
|
|
With diarize=True (a tinydiarize server, see whisper-server-tdrz)
|
|
segments carry .speaker_turn_next and their text is NOT used for
|
|
labeling — use diarize_merge() for that.
|
|
"""
|
|
|
|
def __init__(self, server_url: str, diarize: bool = False):
|
|
self.server_url = server_url.rstrip("/")
|
|
self.diarize = diarize
|
|
self.model_name = None # the server owns the model
|
|
|
|
def transcribe(self, audio, beam_size=5, vad_filter=False,
|
|
language=None, vad_parameters=None, timeout=600):
|
|
with tempfile.TemporaryDirectory(prefix="meetrec-") as td:
|
|
if isinstance(audio, (str, os.PathLike)):
|
|
wav = str(audio) # server resamples to 16 kHz itself
|
|
else:
|
|
wav = os.path.join(td, "in.wav")
|
|
write_wav_float(wav, audio)
|
|
files = [("file", "audio.wav", open(wav, "rb"))]
|
|
fields = {"response_format": "verbose_json",
|
|
"language": language or "auto"}
|
|
if self.diarize:
|
|
fields["tinydiarize"] = "true"
|
|
try:
|
|
body = _multipart_body(fields, files)
|
|
finally:
|
|
files[0][2].close()
|
|
req = urllib.request.Request(
|
|
self.server_url + "/inference", data=body,
|
|
headers={"Content-Type":
|
|
"multipart/form-data; boundary=" + MULTIPART_BOUNDARY})
|
|
try:
|
|
with urllib.request.urlopen(req, timeout=timeout) as r:
|
|
data = json.load(r)
|
|
except urllib.error.HTTPError as e:
|
|
raise RuntimeError(f"server returned {e.code}: "
|
|
f"{e.read()[:300]!r}") from None
|
|
except urllib.error.URLError as e:
|
|
raise RuntimeError(f"cannot reach whisper server at "
|
|
f"{self.server_url}: {e.reason}") from None
|
|
|
|
segs = [SimpleNamespace(start=s["start"], end=s["end"],
|
|
text=s["text"].strip(),
|
|
speaker_turn_next=bool(s.get("speaker_turn_next", False)))
|
|
for s in data.get("segments", [])
|
|
if s.get("text", "").strip()]
|
|
info = SimpleNamespace(
|
|
language=data.get("language") or language or "unknown",
|
|
language_probability=1.0, # not reported by verbose_json
|
|
duration=data.get("duration", 0.0),
|
|
)
|
|
return segs, info
|
|
|
|
|
|
def diarize_merge(segments, turn_segments):
|
|
"""Label segments with alternating "Sprecher 1/2:" using the TURN
|
|
TIMES of a tinydiarize pass; segments in which a turn falls are split.
|
|
Input is returned unchanged when no turns were detected (never
|
|
mislabels). Mirrors the Android app's Diarization object."""
|
|
turns = sorted(t.end for t in turn_segments
|
|
if getattr(t, "speaker_turn_next", False))
|
|
if not turns:
|
|
return list(segments)
|
|
|
|
def seg(a, b, text):
|
|
return SimpleNamespace(start=a, end=b, text=text)
|
|
|
|
speaker, ti = 1, 0
|
|
out = []
|
|
for s in segments:
|
|
while ti < len(turns) and turns[ti] <= s.start:
|
|
speaker = 3 - speaker
|
|
ti += 1
|
|
start, text = s.start, s.text
|
|
while ti < len(turns) and turns[ti] < s.end:
|
|
t = turns[ti]
|
|
frac = (t - s.start) / max(s.end - s.start, 1e-9)
|
|
cut = int(len(text) * frac)
|
|
out.append(seg(start, t, f"Sprecher {speaker}: {text[:cut].strip()}"))
|
|
speaker = 3 - speaker
|
|
ti += 1
|
|
start, text = t, text[cut:]
|
|
out.append(seg(start, s.end, f"Sprecher {speaker}: {text.strip()}"))
|
|
return out
|
|
|
|
|
|
def upload_recording(library_url: str, wav_path: str, out_stem: str,
|
|
duration: float, language, device: str, agenda) -> str:
|
|
"""Upload a recording bundle to the meetrec-server library;
|
|
returns the new recording id. The server then generates the summary
|
|
and agenda coverage automatically (Ollama)."""
|
|
files = []
|
|
for ext in ("wav", "txt", "srt", "json"):
|
|
p = wav_path if ext == "wav" else out_stem + "." + ext
|
|
if os.path.isfile(p):
|
|
files.append((ext, "meeting." + ext, open(p, "rb")))
|
|
started = time.strftime("%Y%m%d-%H%M%S",
|
|
time.localtime(os.path.getmtime(wav_path)))
|
|
fields = {
|
|
"started_at": started,
|
|
"duration_ms": int(duration * 1000),
|
|
"language": language or "",
|
|
"device": device,
|
|
"agenda": json.dumps(list(agenda or []), ensure_ascii=False),
|
|
}
|
|
try:
|
|
body = _multipart_body(fields, files)
|
|
finally:
|
|
for _, _, fh in files:
|
|
fh.close()
|
|
req = urllib.request.Request(
|
|
library_url.rstrip("/") + "/api/recordings", data=body, method="POST",
|
|
headers={"Content-Type": "multipart/form-data; boundary=" + MULTIPART_BOUNDARY})
|
|
try:
|
|
with urllib.request.urlopen(req, timeout=300) as r:
|
|
return json.load(r)["id"]
|
|
except urllib.error.HTTPError as e:
|
|
raise RuntimeError(f"library upload failed: {e.code} "
|
|
f"{e.read()[:200]!r}") from None
|
|
except urllib.error.URLError as e:
|
|
raise RuntimeError(f"cannot reach library at {library_url}: "
|
|
f"{e.reason}") from None
|
|
|
|
|
|
def load_engine(model_name, engine="faster-whisper",
|
|
device="auto", compute_type="int8", server_url=None):
|
|
"""Create a transcription engine. See the Engines section above."""
|
|
if engine == "whisper-cpp":
|
|
return WhisperCppEngine(model_name)
|
|
if engine == "whisper-server":
|
|
url = server_url or os.environ.get("MEETREC_SERVER_URL")
|
|
if not url:
|
|
raise RuntimeError("no server URL: pass --server-url or set "
|
|
"MEETREC_SERVER_URL (e.g. http://100.103.83.12:8085)")
|
|
return WhisperServerEngine(url)
|
|
if engine == "faster-whisper":
|
|
from faster_whisper import WhisperModel
|
|
return WhisperModel(model_name, device=device,
|
|
compute_type=compute_type)
|
|
raise RuntimeError(f"unknown engine {engine!r}; use 'faster-whisper' "
|
|
"or 'whisper-cpp'")
|
|
|
|
|
|
# --------------------------------------------------------------------------
|
|
# Main
|
|
# --------------------------------------------------------------------------
|
|
|
|
def main():
|
|
ap = argparse.ArgumentParser(
|
|
description="Record a meeting and transcribe it with Whisper "
|
|
"(PipeWire / ALSA via PortAudio).")
|
|
ap.add_argument("-s", "--source", default="default",
|
|
help="input to record: 'default' or a substring of a "
|
|
"device name (see --list-sources)")
|
|
ap.add_argument("-m", "--model", default="small",
|
|
help="Whisper model: tiny/base/small/medium/large-v3 "
|
|
"(default: small)")
|
|
ap.add_argument("-o", "--output", default="meeting",
|
|
help="output file stem (default: meeting)")
|
|
ap.add_argument("--live", type=int, default=8, metavar="SEC",
|
|
help="live transcription interval in seconds; 0 disables "
|
|
"(default: 8)")
|
|
ap.add_argument("--language", default=None,
|
|
help="force language code, e.g. en, de (default: autodetect)")
|
|
ap.add_argument("--engine", default="faster-whisper",
|
|
choices=["faster-whisper", "whisper-cpp", "whisper-server"],
|
|
help="transcription backend (default: faster-whisper). "
|
|
"whisper-cpp can use a GPU via Vulkan; whisper-server "
|
|
"transcribes on a remote whisper.cpp server")
|
|
ap.add_argument("--server-url", default=None,
|
|
help="whisper-server base URL, e.g. http://100.103.83.12:8085 "
|
|
"(or set MEETREC_SERVER_URL); whisper-server engine only")
|
|
ap.add_argument("--device", default="auto",
|
|
help="compute device: auto / cpu / cuda (default: auto); "
|
|
"faster-whisper only")
|
|
ap.add_argument("--compute-type", default="int8",
|
|
help="int8 / int8_float16 / float16 / float32 "
|
|
"(default: int8); faster-whisper only")
|
|
ap.add_argument("--list-sources", action="store_true",
|
|
help="list available input devices and exit")
|
|
ap.add_argument("--agenda-file", default=None, metavar="PATH",
|
|
help="text file with agenda items, one per line — "
|
|
"uploaded with the recording for the coverage "
|
|
"check (see --library-url)")
|
|
ap.add_argument("--diarize-url", default=os.environ.get("MEETREC_DIARIZE_URL"),
|
|
help="tinydiarize whisper-server URL for speaker labels "
|
|
"(Sprecher 1/2), e.g. http://100.103.83.12:8086 "
|
|
"(or set MEETREC_DIARIZE_URL)")
|
|
ap.add_argument("--library-url", default=os.environ.get("MEETREC_LIBRARY_URL"),
|
|
help="meetrec-server URL; uploads the recording after "
|
|
"transcription and triggers summary + agenda check "
|
|
"on the server (or set MEETREC_LIBRARY_URL)")
|
|
args = ap.parse_args()
|
|
|
|
if args.list_sources:
|
|
list_sources()
|
|
return
|
|
|
|
device = resolve_source(args.source)
|
|
|
|
if args.engine == "whisper-server":
|
|
print(f"using whisper server at "
|
|
f"{args.server_url or os.environ.get('MEETREC_SERVER_URL')} "
|
|
f"(model lives on the server)")
|
|
else:
|
|
print(f"loading Whisper model {args.model!r} via {args.engine!r} "
|
|
f"(first run downloads it to ~/.cache)...")
|
|
try:
|
|
model = load_engine(args.model, engine=args.engine,
|
|
device=args.device,
|
|
compute_type=args.compute_type,
|
|
server_url=args.server_url)
|
|
except Exception as e:
|
|
print(f"error: {e}", file=sys.stderr)
|
|
sys.exit(1)
|
|
|
|
rec = Recorder(device, args.output + ".wav")
|
|
rec.start()
|
|
print(f"recording to {args.output}.wav at {rec.rate} Hz "
|
|
f"(Ctrl+C to stop)", flush=True)
|
|
|
|
stop = threading.Event()
|
|
worker = None
|
|
if args.live > 0:
|
|
worker = threading.Thread(
|
|
target=live_worker, args=(model, rec, args,
|
|
args.output + ".live.srt", stop),
|
|
daemon=True)
|
|
worker.start()
|
|
|
|
try:
|
|
while not stop.is_set():
|
|
stop.wait(0.5)
|
|
except KeyboardInterrupt:
|
|
print("\nstopping...", flush=True)
|
|
finally:
|
|
stop.set()
|
|
rec.stop()
|
|
if worker:
|
|
worker.join(timeout=60)
|
|
agenda = []
|
|
if args.agenda_file:
|
|
with open(args.agenda_file) as f:
|
|
agenda = [ln.strip() for ln in f if ln.strip()]
|
|
final_transcribe(model, args.output + ".wav", args.output, args.language,
|
|
diarize_url=args.diarize_url, agenda=agenda,
|
|
library_url=args.library_url)
|
|
|
|
|
|
if __name__ == "__main__":
|
|
main()
|