chore: snapshot livekit-cameras after serial pin + speechbrain lengths fix
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Hermes
+225
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"""One camera -> one LiveKit participant.
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Run as a subprocess per camera: python -m cameras.run_worker --index 0 ...
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"""
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from __future__ import annotations
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import argparse
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import asyncio
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import contextlib
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import json
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import logging
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import os
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import signal
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import subprocess
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import sys
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import time
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import numpy as np
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log = logging.getLogger("cameras.worker")
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def main() -> int:
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ap = argparse.ArgumentParser()
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ap.add_argument("--camera-json", required=True,
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help="JSON of one discovered camera dict")
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ap.add_argument("--room", required=True)
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ap.add_argument("--url", required=True)
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ap.add_argument("--token", required=True)
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ap.add_argument("--width", type=int, default=1280)
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ap.add_argument("--height", type=int, default=720)
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ap.add_argument("--fps", type=int, default=30)
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ap.add_argument("--audio-rate", type=int, default=16000)
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ap.add_argument("--enhance-mode", default="auto")
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ap.add_argument("--enhance-model",
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default="speechbrain/metricgan-plus-voicebank")
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ap.add_argument("--enhance-chunk-s", type=float, default=0.2)
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args = ap.parse_args()
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cam = json.loads(args.camera_json)
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logging.basicConfig(
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level=logging.INFO,
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format=f"%(asctime)s [{cam['identity']}] %(levelname)s %(message)s",
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)
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from config import Config # noqa: F401 (ensures .env is loaded)
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import livekit.rtc as rtc
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from audio_cleanup import AudioCleaner
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identity = cam["identity"]
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video_dev = cam["video_device"]
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audio_card = cam.get("audio_card")
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async def run() -> int:
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cleaner = AudioCleaner(
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mode=args.enhance_mode,
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model_source=args.enhance_model,
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sample_rate=args.audio_rate,
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)
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log.info("audio backend: %s", cleaner.backend)
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room = rtc.Room()
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await room.connect(args.url, args.token)
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log.info("connected to room %r as %r", room.name, identity)
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# ---------------- video ----------------
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video_source = rtc.VideoSource(args.width, args.height)
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video_track = rtc.LocalVideoTrack.create_video_track(
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f"{identity}-video", video_source)
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vproc = subprocess.Popen(
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["ffmpeg", "-hide_banner", "-loglevel", "error",
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"-f", "v4l2",
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"-input_format", "mjpeg",
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"-video_size", f"{args.width}x{args.height}",
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"-framerate", str(args.fps),
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"-i", video_dev,
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"-f", "rawvideo", "-pix_fmt", "bgra",
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"pipe:1"],
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stdout=subprocess.PIPE,
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)
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frame_bytes = args.width * args.height * 4
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video_frames = 0
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async def video_loop():
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nonlocal video_frames
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loop = asyncio.get_running_loop()
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last = time.monotonic()
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while vproc.poll() is None:
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data = await loop.run_in_executor(
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None, vproc.stdout.read, frame_bytes)
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if not data or len(data) < frame_bytes:
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if not data:
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break
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# partial read: accumulate (v4l2/ffmpeg shouldn't do this)
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keep = b""
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while len(keep) < frame_bytes:
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more = await loop.run_in_executor(
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None, vproc.stdout.read,
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frame_bytes - len(keep))
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if not more:
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return
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keep += more
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data = keep
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now = time.monotonic()
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ts_us = int((now - last) * 1_000_000)
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try:
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video_source.capture_frame(
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rtc.VideoFrame(
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args.width, args.height,
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rtc.VideoBufferType.BGRA, data),
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timestamp_us=ts_us,
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)
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video_frames += 1
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except Exception: # noqa: BLE001
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log.exception("video capture_frame failed")
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# ---------------- audio ----------------
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aproc: subprocess.Popen | None = None
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atrack: rtc.LocalAudioTrack | None = None
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if audio_card is not None:
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audio_source = rtc.AudioSource(
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sample_rate=args.audio_rate, num_channels=1)
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atrack = rtc.LocalAudioTrack.create_audio_track(
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f"{identity}-audio", audio_source)
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chunk_s = args.enhance_chunk_s
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chunk = int(chunk_s * args.audio_rate) # samples per channel
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async def audio_loop():
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assert aproc is not None and atrack is not None
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loop = asyncio.get_running_loop()
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# mic delivers stereo s16le: bytes per chunk =
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# chunk * 2 ch * 2 bytes
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need = chunk * 2 * 2
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buf = b""
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while aproc.poll() is None:
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data = await loop.run_in_executor(
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None, aproc.stdout.read, 4096)
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if not data:
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break
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buf += data
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while len(buf) >= need:
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piece, buf = buf[:need], buf[need:]
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pcm_stereo = np.frombuffer(piece, dtype=np.int16)
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pcm_mono = pcm_stereo.reshape(-1, 2).mean(
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axis=1).astype(np.int16)
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# align to full chunks
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usable = (len(pcm_mono) // chunk) * chunk
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if usable:
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cleaned = await loop.run_in_executor(
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None, cleaner.process,
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pcm_mono[:usable])
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await audio_source.capture_frame(
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rtc.AudioFrame(
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cleaned.tobytes(),
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args.audio_rate,
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1,
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len(cleaned)))
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aproc = subprocess.Popen(
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["ffmpeg", "-hide_banner", "-loglevel", "error",
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"-f", "alsa", "-i", f"hw:{audio_card},0",
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"-ar", str(args.audio_rate), "-ac", "2",
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"-f", "s16le", "pipe:1"],
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stdout=subprocess.PIPE,
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)
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# ---------------- publish ----------------
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from livekit.rtc import TrackSource, TrackPublishOptions
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pub_opts = TrackPublishOptions(
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source=TrackSource.SOURCE_CAMERA, simulcast=False)
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await room.local_participant.publish_track(video_track, pub_opts)
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if atrack is not None:
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await room.local_participant.publish_track(
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atrack,
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TrackPublishOptions(
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source=TrackSource.SOURCE_MICROPHONE,
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dtx=False))
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log.info("published video+audio (audio backend: %s)",
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cleaner.backend)
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else:
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log.info("published video only (no audio card matched)")
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log.info("status: UP identity=%s video=%s audio=%s",
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identity, video_dev,
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f"hw:{audio_card},0" if audio_card is not None else "none")
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# ---------------- run ----------------
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tasks = [asyncio.create_task(video_loop(), name="video")]
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if aproc is not None:
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tasks.append(asyncio.create_task(audio_loop(), name="audio"))
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stop = asyncio.Event()
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loop = asyncio.get_running_loop()
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for sig in (signal.SIGTERM, signal.SIGINT):
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with contextlib.suppress(NotImplementedError):
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loop.add_signal_handler(sig, stop.set)
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await stop.wait()
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for t in tasks:
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t.cancel()
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await asyncio.gather(*tasks, return_exceptions=True)
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for p in (vproc, aproc):
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if p is not None and p.poll() is None:
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with contextlib.suppress(Exception):
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p.terminate()
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with contextlib.suppress(Exception):
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p.wait(timeout=3)
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log.info("stopped; delivered %d video frames", video_frames)
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with contextlib.suppress(Exception):
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await room.disconnect()
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return 0
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try:
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return asyncio.run(run())
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except Exception: # noqa: BLE001
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log.exception("worker fatal")
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return 1
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if __name__ == "__main__":
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sys.exit(main())
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