somewhat stable
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@@ -22,7 +22,7 @@ import socket
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import time
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from pathlib import Path
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from queue import Queue, Empty
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from threading import Thread
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from threading import Thread, Event
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from cht.config import (
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AUDIO_EXTRACT_INTERVAL,
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@@ -43,7 +43,7 @@ class SessionProcessor:
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self.frames_dir = session_dir / "frames"
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self.audio_dir = session_dir / "audio"
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self._stop_flags: set[str] = set()
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self._stop_event = Event()
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self._threads: dict[str, Thread] = {}
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self._on_new_frames = None
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self._on_new_audio = None
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@@ -289,26 +289,26 @@ class SessionProcessor:
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socket_path = DATA_DIR / "scene.sock"
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# Wait for the socket to appear (server creates it on session start).
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while "stop" not in self._stop_flags:
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while not self._stop_event.is_set():
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if socket_path.exists():
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break
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time.sleep(0.5)
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if "stop" in self._stop_flags:
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if self._stop_event.is_set():
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return
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while "stop" not in self._stop_flags:
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while not self._stop_event.is_set():
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try:
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self._run_scene_session(socket_path, threshold)
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except Exception:
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log.exception("Scene detector error")
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if "stop" in self._stop_flags:
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if self._stop_event.is_set():
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break
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# If the socket is gone, the session ended — don't retry.
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if not socket_path.exists():
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log.info("Scene detector: socket gone, session ended")
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break
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log.info("Scene detector: reconnecting in 2s...")
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time.sleep(2.0)
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self._stop_event.wait(timeout=2.0)
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log.info("Scene detector stopped")
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@@ -333,7 +333,7 @@ class SessionProcessor:
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# Thread: socket → ffmpeg stdin
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def _feed_stdin():
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try:
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while "stop" not in self._stop_flags:
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while not self._stop_event.is_set():
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data = sock.recv(65536)
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if not data:
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break
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@@ -411,10 +411,16 @@ class SessionProcessor:
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self._threads["audio_extractor"] = t
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def stop(self):
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self._stop_flags.add("stop")
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self._stop_event.set()
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for name, proc in getattr(self, "_procs", {}).items():
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ff.stop_proc(proc, timeout=3)
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self._procs = {} if hasattr(self, "_procs") else {}
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self._procs = {}
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# Join all threads so caller knows they're done before starting a new session
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for name, t in list(self._threads.items()):
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t.join(timeout=5)
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if t.is_alive():
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log.warning("Thread %s still alive after stop timeout", name)
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self._threads.clear()
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def _has_audio_stream(self, seg: Path) -> bool:
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try:
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@@ -441,9 +447,7 @@ class SessionProcessor:
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chunk_num = 0
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current_source = None
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while "stop" not in self._stop_flags:
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time.sleep(AUDIO_EXTRACT_INTERVAL)
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while not self._stop_event.wait(timeout=AUDIO_EXTRACT_INTERVAL):
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source = self._find_audio_source()
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if not source:
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continue
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@@ -461,6 +465,16 @@ class SessionProcessor:
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if safe_duration is None or safe_duration <= 0:
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continue
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# Fail-safe: processed_time can accumulate past the file if the
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# source was recreated (e.g. server restarted same session).
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if processed_time > safe_duration:
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log.warning(
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"Audio extractor: processed_time %.1fs > file duration %.1fs — resetting",
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processed_time, safe_duration,
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)
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processed_time = 0.0
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chunk_num = 0
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process_to = safe_duration - AUDIO_SAFETY_MARGIN
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if process_to <= processed_time + 1.0:
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continue
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@@ -147,7 +147,7 @@ class MonitorWidget(Gtk.Box):
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gl_area.make_current()
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self._live_player = Player()
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self._live_player.init_gl(
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update_callback=lambda: GLib.idle_add(self._live_gl.queue_render)
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update_callback=lambda: GLib.idle_add(self._live_gl.queue_render, priority=GLib.PRIORITY_HIGH)
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)
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log.info("Live player created")
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if self._live_source_url and not self._live_loaded:
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@@ -162,7 +162,7 @@ class MonitorWidget(Gtk.Box):
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self._live_loaded = False
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def _on_live_render(self, gl_area, _ctx):
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if not self._live_player:
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if not self._live_player or not self._live_loaded:
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return True
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fbo = ctypes.c_int(0)
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_libGL.glGetIntegerv(GL_DRAW_FRAMEBUFFER_BINDING, ctypes.byref(fbo))
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@@ -175,7 +175,7 @@ class MonitorWidget(Gtk.Box):
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gl_area.make_current()
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self._review_player = Player()
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self._review_player.init_gl(
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update_callback=lambda: GLib.idle_add(self._review_gl.queue_render)
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update_callback=lambda: GLib.idle_add(self._review_gl.queue_render, priority=GLib.PRIORITY_HIGH)
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)
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log.info("Review player created")
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@@ -103,12 +103,9 @@ class Player:
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def load_live(self, url):
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"""Load a live stream URL with low-latency options."""
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self._player["cache"] = "no"
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self._player["demuxer-max-bytes"] = "256KiB"
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self._player["demuxer-readahead-secs"] = 0
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self._player["demuxer-max-bytes"] = "512KiB"
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self._player["demuxer-readahead-secs"] = 0.5
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self._player["audio-buffer"] = 0.1
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self._player["untimed"] = True
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self._player["video-latency-hacks"] = True
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self._player["interpolation"] = False
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log.info("mpv load_live: %s", url)
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self._player.loadfile(str(url), mode="replace")
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@@ -508,7 +508,8 @@ class ChtWindow(Adw.ApplicationWindow):
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self._connect_btn.add_css_class("suggested-action")
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if reload_session and last_session_id:
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# Transition to review mode — _load_session handles UI setup
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# Stop live player before transitioning to review mode
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self._monitor.reset()
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self._load_session(last_session_id)
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return
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@@ -1,13 +1,16 @@
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//! Subprocess backend: spawn ffmpeg CLI for capture+encode.
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//!
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//! Spawns ffmpeg with the same hardware pipeline as `stream_av.sh`:
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//! kmsgrab → hwmap=derive_device=vaapi → scale_vaapi → h264_vaapi
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//! kmsgrab -vblank_source vsync → hwmap=derive_device=vaapi → scale_vaapi → h264_vaapi
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//! + PulseAudio desktop audio + mic → amix → AAC
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//!
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//! -vblank_source vsync forces a frame grab on every display vblank, regardless
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//! of page flips. Without it, kmsgrab only grabs when the compositor flips a
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//! new buffer — a static/slow screen yields 1fps.
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//!
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//! ffmpeg outputs NUT format to stdout. We demux that pipe with ffmpeg-next
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//! to get proper AVPackets (keyframe flags, timestamps) without parsing
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//! bytestreams. NUT is lighter than mpegts — no TS overhead, exact packet
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//! metadata in the container layer.
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//! bytestreams.
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use std::os::fd::AsRawFd;
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use std::os::unix::io::RawFd;
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@@ -76,8 +79,6 @@ pub fn run(
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while !stop_watcher.load(Ordering::Relaxed) {
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std::thread::sleep(std::time::Duration::from_millis(100));
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}
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// Send SIGINT to ffmpeg so it flushes and closes stdout,
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// which unblocks the packet iterator in demux_and_send.
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use nix::sys::signal::{kill, Signal};
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use nix::unistd::Pid;
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let _ = kill(Pid::from_raw(child_pid as i32), Signal::SIGINT);
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@@ -148,6 +149,9 @@ fn detect_default_source(pulse_server: &str) -> Option<String> {
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fn spawn_ffmpeg(cfg: &SubprocessConfig) -> Result<Child> {
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let audio = detect_audio_sources();
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// fps filter after scale_vaapi pads/duplicates frames to fill gaps when
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// kmsgrab captures fewer frames than the target rate (e.g. compositor
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// skips flips on static content). Keeps the output stream at a stable fps.
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let filter = format!(
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"hwmap=derive_device=vaapi,scale_vaapi=w={}:h={}:format=nv12,fps={}",
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cfg.width, cfg.height, cfg.fps,
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@@ -158,6 +162,9 @@ fn spawn_ffmpeg(cfg: &SubprocessConfig) -> Result<Child> {
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"-init_hw_device".into(), format!("drm=drm:{}", cfg.device),
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"-init_hw_device".into(), "vaapi=va@drm".into(),
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// Video input (kmsgrab)
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// -vblank_source vsync: grab on every display vblank, not just page flips.
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// Without this, a static screen (e.g. talking-head meeting) gives 1fps
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// because the compositor rarely flips a new buffer.
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"-thread_queue_size".into(), "512".into(),
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"-device".into(), cfg.device.clone(),
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"-f".into(), "kmsgrab".into(),
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@@ -1,7 +1,7 @@
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#!/bin/bash
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# Build and run the media client (sender)
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# Requires DRM master access — runs under sudo unless already root.
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# Usage: ./client.sh [server_addr] e.g. ./client.sh mcrndeb:4444
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# Usage: ./client.sh [server_addr] e.g. ./client.sh mcrndeb:4447
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set -euo pipefail
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MEDIA_DIR="$(cd "$(dirname "$0")/.." && pwd)"
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