From 2544c7cb78d458450dfc39fc01930abff3dd2b27 Mon Sep 17 00:00:00 2001
From: =?UTF-8?q?Olivier=20Cr=C3=AAte?= <olivier.crete@collabora.com>
Date: Tue, 3 Jan 2023 17:44:37 -0500
Subject: [PATCH] webrtc examples: Force regular non-MULTIOPUS

Using MULTIOPUS breaks with most browsers

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3685>
---
 .../webrtc/multiparty-sendrecv/gst-rust/src/main.rs         | 2 +-
 .../webrtc/sendonly/webrtc-unidirectional-h264.c            | 3 ++-
 .../gst-examples/webrtc/sendrecv/gst-rust/src/main.rs       | 2 +-
 .../gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c      | 6 +++---
 4 files changed, 7 insertions(+), 6 deletions(-)

diff --git a/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs b/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs
index ddc61a0c3c..a8e373582b 100644
--- a/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs
+++ b/subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs
@@ -160,7 +160,7 @@ impl App {
             &format!(
                 "videotestsrc is-live=true ! vp8enc deadline=1 ! rtpvp8pay pt=96 ! tee name=video-tee ! \
                  queue ! fakesink sync=true \
-                 audiotestsrc wave=ticks is-live=true ! opusenc ! rtpopuspay pt=97 ! tee name=audio-tee ! \
+                 audiotestsrc wave=ticks is-live=true ! opusenc ! rtpopuspay pt=97 ! application/x-rtp,encoding-name=OPUS ! tee name=audio-tee ! \
                  queue ! fakesink sync=true \
                  audiotestsrc wave=silence is-live=true ! audio-mixer. \
                  audiomixer name=audio-mixer sink_0::mute=true ! audioconvert ! audioresample ! autoaudiosink \
diff --git a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
index 75f4a3e56c..42cd9d2d60 100644
--- a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
+++ b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
@@ -244,7 +244,8 @@ create_receiver_entry (SoupWebsocketConnection * connection)
       "application/x-rtp,media=video,encoding-name=H264,payload="
       RTP_PAYLOAD_TYPE " ! webrtcbin. "
       "autoaudiosrc is-live=1 ! queue max-size-buffers=1 leaky=downstream ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
-      RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error);
+      RTP_AUDIO_PAYLOAD_TYPE
+      " ! application/x-rtp,encoding-name=OPUS ! webrtcbin. ", &error);
   if (error != NULL) {
     g_error ("Could not create WebRTC pipeline: %s\n", error->message);
     g_error_free (error);
diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs b/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
index a594e5071c..836ab5d564 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
@@ -115,7 +115,7 @@ impl App {
         // Create the GStreamer pipeline
         let pipeline = gst::parse_launch(
         "videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 ! rtpvp8pay pt=96 ! webrtcbin. \
-         audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=97 ! webrtcbin. \
+         audiotestsrc is-live=true ! opusenc ! rtpopuspay pt=97 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \
          webrtcbin name=webrtcbin"
     )?;
 
diff --git a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
index edbaafac2a..a201c7473e 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
@@ -445,9 +445,9 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
        * fixes stuttery video playback in Chrome */
       "rtpvp8pay name=videopay picture-id-mode=15-bit ! "
       "queue ! %s,payload=%u ! sendrecv. "
-      "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay ! "
-      "queue ! %s,payload=%u ! sendrecv. ", RTP_CAPS_VP8, vp8_pt,
-      RTP_CAPS_OPUS, opus_pt);
+      "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay name=audiopay "
+      " ! application/x-rtp, encoding-name=OPUS ! queue ! %s,payload=%u ! sendrecv. ",
+      RTP_CAPS_VP8, vp8_pt, RTP_CAPS_OPUS, opus_pt);
 
   pipe1 = gst_parse_launch (pipeline, &error);
   g_free (pipeline);