From e471bd02b265604e060f030200fb1295e3c1b412 Mon Sep 17 00:00:00 2001
From: Zac Gaetano <zgaetano@wilddragon.net>
Date: Fri, 17 Apr 2026 08:51:22 -0400
Subject: [PATCH] test(webrtc): add FFmpeg publisher script for M1 PoC

Generates a synthetic testsrc2 video + sine audio and pushes H.264/Opus
RTP to the webrtc-poc's UDP port, using the hard-coded payload types
(102 video, 111 audio) the M1 forwarder dispatches on. Intended to be
run alongside test/whep-client (M1 Task 11) for end-to-end verification.
---
 test/publish.sh | 46 ++++++++++++++++++++++++++++++++++++++++++++++
 1 file changed, 46 insertions(+)
 create mode 100755 test/publish.sh

diff --git a/test/publish.sh b/test/publish.sh
new file mode 100755
index 0000000..8e59ede
--- /dev/null
+++ b/test/publish.sh
@@ -0,0 +1,46 @@
+#!/usr/bin/env bash
+# publish.sh — M1 PoC test publisher.
+#
+# Generates a synthetic test pattern + sine tone, encodes to H.264
+# baseline (PT=102) and Opus (PT=111), then sends both RTP streams
+# muxed on a single UDP port to match the M1 webrtc-poc server, which
+# reads one UDP port and dispatches by payload type.
+#
+# Usage:
+#   ./test/publish.sh [host] [port]
+#
+# Defaults: host=127.0.0.1 port=10000
+set -euo pipefail
+
+HOST="${1:-127.0.0.1}"
+PORT="${2:-10000}"
+
+echo "publishing test pattern + tone to rtp://${HOST}:${PORT}"
+echo "video PT=102 (H.264 baseline), audio PT=111 (Opus)"
+echo "press Ctrl-C to stop."
+
+# -re                   real-time pace (wall-clock)
+# testsrc / sine        synthetic A/V so no devices needed
+# libx264 baseline      widely compatible profile for WebRTC
+# -tune zerolatency     minimize encoder buffering
+# -bsf:v h264_mp4toannexb  ensure Annex-B for RTP packetization
+# -payload_type 102/111 match the hard-coded PTs in forward.go
+# -f rtp_mpegts fails   (we need plain rtp, not mpegts-in-rtp)
+# Using two separate -f rtp outputs, both to the same UDP port.
+# FFmpeg 4.x requires an SDP file per output; we write them to /tmp
+# but the server doesn't use them — it only cares about PT.
+exec ffmpeg -hide_banner -loglevel warning \
+  -re \
+  -f lavfi -i "testsrc2=size=640x360:rate=30" \
+  -f lavfi -i "sine=frequency=440:sample_rate=48000" \
+  -map 0:v:0 \
+    -c:v libx264 -profile:v baseline -preset veryfast -tune zerolatency \
+    -g 30 -keyint_min 30 -x264-params "repeat-headers=1" \
+    -pix_fmt yuv420p \
+    -bsf:v h264_mp4toannexb \
+    -payload_type 102 \
+    -f rtp "rtp://${HOST}:${PORT}?pkt_size=1200" \
+  -map 1:a:0 \
+    -c:a libopus -b:a 64k -ar 48000 -ac 2 \
+    -payload_type 111 \
+    -f rtp "rtp://${HOST}:${PORT}?pkt_size=1200"