From b7c0e8bc844c05bee02d85075529411f44f4fec5 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/3675>
---
 .../webrtc/multiparty-sendrecv/gst-rust/src/main.rs         | 2 +-
 .../webrtc/sendonly/webrtc-unidirectional-h264.c            | 6 ++++--
 .../gst-examples/webrtc/sendrecv/gst-rust/src/main.rs       | 2 +-
 .../gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c      | 3 ++-
 4 files changed, 8 insertions(+), 5 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 8e84f9a757..b00cda9068 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
@@ -159,7 +159,7 @@ 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 1fedf8494c..7df8bd9c74 100644
--- a/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
+++ b/subprojects/gst-examples/webrtc/sendonly/webrtc-unidirectional-h264.c
@@ -248,8 +248,10 @@ create_receiver_entry (SoupWebsocketConnection * connection)
       "rtph264pay config-interval=-1 name=payloader aggregate-mode=zero-latency ! "
       "application/x-rtp,media=video,encoding-name=H264,payload="
       RTP_PAYLOAD_TYPE " ! webrtcbin. "
-      "autoaudiosrc ! queue max-size-buffers=1 leaky=downstream ! audioconvert ! audioresample ! opusenc ! rtpopuspay pt="
-      RTP_AUDIO_PAYLOAD_TYPE " ! webrtcbin. ", &error);
+      "autoaudiosrc ! queue max-size-buffers=1 leaky=downstream"
+      " ! audioconvert ! audioresample ! opusenc  ! rtpopuspay pt="
+      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 585eb17f09..c06d777906 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
@@ -116,7 +116,7 @@ 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 9455e26273..d1da1155d0 100644
--- a/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst/webrtc-sendrecv.c
@@ -481,7 +481,8 @@ start_pipeline (gboolean create_offer, guint opus_pt, guint vp8_pt)
   audio_desc =
       g_strdup_printf
       ("audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample"
-      "! queue ! opusenc ! rtpopuspay name=audiopay pt=%u ! queue", opus_pt);
+      "! queue ! opusenc ! rtpopuspay name=audiopay pt=%u "
+      "! application/x-rtp, encoding-name=OPUS ! queue", opus_pt);
   audio_bin = gst_parse_bin_from_description (audio_desc, TRUE, &audio_error);
   g_free (audio_desc);
   if (audio_error) {