Audio capture and streaming app for webOS 5/6

Captures the TV's audio and sends it out over several transports. The
primary one is HyperHDR: RTP/L16 to a host-side loopback device, since
HyperHDR has no network audio input of its own. A second route renders
the spectrum on the TV and sends FlatBuffers images to port 19400
instead, for setups where touching the host's sound config is not an
option.

  native/       the service: capture backends (PulseAudio, ALSA, exec,
                test tone, all dlopen-based), DSP, and one file per sink
  frontend/     D-pad driven UI at a fixed 1920x1080
  servicefiles/ native service manifest plus the boot script
  host/         RTP receiver and the loopback installer for the HyperHDR
                machine
  tools/        build/package, asset generation, Homebrew Channel
                manifest, on-TV probe
  test/         host-side suites: FlatBuffers and RTP verified against
                real decoders, the engine end to end, the page in jsdom

Co-Authored-By: Claude Opus 5 <noreply@anthropic.com>

tools/tv-probe.sh

@@ -0,0 +1,143 @@
+#!/bin/sh
+# Reports what audio a rooted webOS TV actually exposes.
+#
+# Which capture backend works depends on the model and firmware: some sets run
+# PulseAudio with a monitor source, some only offer ALSA, some need an external
+# helper. Run this once on the TV and the answer is usually obvious.
+#
+# On the TV:
+#     sh tv-probe.sh
+#
+# From here, over the Homebrew Channel's ssh:
+#     ssh -p 9922 root@TV-IP 'sh -s' < tools/tv-probe.sh
+#     ares-shell --device tv -r "$(cat tools/tv-probe.sh)"
+#
+# Reads only. Nothing here changes the TV.
+
+header() {
+  printf '\n=== %s\n' "$1"
+}
+
+have() {
+  command -v "$1" >/dev/null 2>&1
+}
+
+show() {
+  # show <label> <file>
+  if [ -r "$2" ]; then
+    printf -- '--- %s\n' "$1"
+    cat "$2"
+  fi
+}
+
+header "Identity"
+printf 'uid:      %s\n' "$(id -u 2>/dev/null)"
+printf 'kernel:   %s\n' "$(uname -r 2>/dev/null)"
+printf 'machine:  %s\n' "$(uname -m 2>/dev/null)"
+for f in /etc/starfish-release /etc/os-release; do
+  show "$f" "$f"
+done
+if [ -r /var/run/nyx/device_info.json ]; then
+  printf -- '--- device info\n'
+  cat /var/run/nyx/device_info.json
+  printf '\n'
+fi
+
+header "Sound devices"
+if [ -d /dev/snd ]; then
+  ls -l /dev/snd
+else
+  echo "no /dev/snd — the ALSA backend cannot work"
+fi
+
+header "ALSA"
+show "cards" /proc/asound/cards
+show "pcm" /proc/asound/pcm
+show "modules" /proc/asound/modules
+if [ -d /proc/asound ]; then
+  for dir in /proc/asound/card*/pcm*c; do
+    [ -d "$dir" ] || continue
+    printf -- '--- capture stream %s\n' "$dir"
+    [ -r "$dir/info" ] && cat "$dir/info"
+  done
+else
+  echo "no /proc/asound — no ALSA on this firmware"
+fi
+
+header "Tools"
+for tool in arecord aplay amixer pactl parec pacat pulseaudio gst-launch-1.0 ffmpeg luna-send; do
+  if have "$tool"; then
+    printf '%-16s %s\n' "$tool" "$(command -v "$tool")"
+  else
+    printf '%-16s -\n' "$tool"
+  fi
+done
+
+header "Libraries"
+# The service dlopen()s these rather than linking them, so what matters is
+# whether the file exists at all and under which soname.
+for pattern in \
+  /usr/lib/libasound.so* /lib/libasound.so* \
+  /usr/lib/libpulse.so* /lib/libpulse.so* \
+  /usr/lib/libpulse-simple.so* /lib/libpulse-simple.so*
+do
+  for lib in $pattern; do
+    [ -e "$lib" ] && ls -l "$lib"
+  done
+done 2>/dev/null
+
+header "PulseAudio"
+ps ax 2>/dev/null | grep -i '[p]ulseaudio' || ps 2>/dev/null | grep -i '[p]ulseaudio' \
+  || echo "no pulseaudio process found"
+printf -- '--- sockets\n'
+for sock in /var/run/pulse/native /run/pulse/native /tmp/pulse-*/native \
+            /var/run/pulse/*.socket /dev/socket/pulse/*
+do
+  [ -e "$sock" ] && ls -l "$sock"
+done 2>/dev/null
+printf -- '--- environment of the audio daemon\n'
+for pid in $(ps ax 2>/dev/null | grep -i '[a]udiod\|[p]ulseaudio' | awk '{print $1}'); do
+  if [ -r "/proc/$pid/environ" ]; then
+    printf 'pid %s: ' "$pid"
+    tr '\0' '\n' < "/proc/$pid/environ" | grep -i 'PULSE\|XDG_RUNTIME' | tr '\n' ' '
+    printf '\n'
+  fi
+done
+
+if have pactl; then
+  printf -- '--- pactl info\n'
+  PULSE_SERVER="${PULSE_SERVER:-unix:/var/run/pulse/native}" pactl info 2>&1 | head -20
+  printf -- '--- sources (monitors are what to capture)\n'
+  PULSE_SERVER="${PULSE_SERVER:-unix:/var/run/pulse/native}" pactl list short sources 2>&1
+fi
+
+if have arecord; then
+  header "ALSA capture devices (arecord -l)"
+  arecord -l 2>&1
+  printf -- '--- PCMs (arecord -L)\n'
+  arecord -L 2>&1 | head -40
+fi
+
+header "What to put in the app"
+cat <<'EOF'
+Look at the output above and set Capture > Backend accordingly:
+
+  PulseAudio   a pulseaudio process and a socket exist, and pactl lists a
+               source ending in ".monitor". Use that name as the Device, or
+               leave Device blank to take the default monitor.
+
+  ALSA         /proc/asound lists a card with a capture PCM. Use hw:X,Y with
+               the card and device numbers from "arecord -l".
+
+  Command      neither works but arecord/parec/gst-launch is present. Set the
+               Command to something that writes raw S16LE to stdout, e.g.
+                 arecord -D hw:0,0 -f S16_LE -r 48000 -c 2 -t raw
+
+  Test tone    nothing at all. Use this to prove the network path works while
+               you keep looking.
+
+If everything is missing, the audio path on this firmware is locked inside the
+proprietary audio daemon. Every sink depends on captured audio, so the test
+tone is all that will run until a capture route is found — use it to prove the
+network side, then come back to this list.
+EOF