From 4df6d21992cd2e69c878266e350417d8c8abc48a Mon Sep 17 00:00:00 2001
From: Mathieu Duponchelle <mathieu@centricular.com>
Date: Mon, 15 Oct 2018 20:45:57 +0200
Subject: [PATCH] sendrecv: port all examples to use a max-bundle policy

---
 webrtc/sendrecv/gst-rust/src/main.rs        | 3 +++
 webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs | 8 ++++----
 webrtc/sendrecv/gst/webrtc-sendrecv.c       | 6 +++---
 webrtc/sendrecv/gst/webrtc-sendrecv.py      | 6 +++---
 4 files changed, 13 insertions(+), 10 deletions(-)

diff --git a/webrtc/sendrecv/gst-rust/src/main.rs b/webrtc/sendrecv/gst-rust/src/main.rs
index 5d6041c604..531a13322d 100644
--- a/webrtc/sendrecv/gst-rust/src/main.rs
+++ b/webrtc/sendrecv/gst-rust/src/main.rs
@@ -316,6 +316,7 @@ fn add_video_source(pipeline: &gst::Pipeline, webrtcbin: &gst::Element) -> Resul
     let vp8enc = gst::ElementFactory::make("vp8enc", None).unwrap();
 
     videotestsrc.set_property_from_str("pattern", "ball");
+    videotestsrc.set_property("is-live", &true).unwrap();
     vp8enc.set_property("deadline", &1i64).unwrap();
 
     let rtpvp8pay = gst::ElementFactory::make("rtpvp8pay", None).unwrap();
@@ -355,6 +356,7 @@ fn add_audio_source(pipeline: &gst::Pipeline, webrtcbin: &gst::Element) -> Resul
     let queue3 = gst::ElementFactory::make("queue", None).unwrap();
 
     audiotestsrc.set_property_from_str("wave", "red-noise");
+    audiotestsrc.set_property("is-live", &true).unwrap();
 
     pipeline.add_many(&[
         &audiotestsrc,
@@ -430,6 +432,7 @@ impl AppControl {
         pipeline.add(&webrtcbin)?;
 
         webrtcbin.set_property_from_str("stun-server", STUN_SERVER);
+        webrtcbin.set_property_from_str("bundle-policy", "max-bundle");
 
         add_video_source(&pipeline, &webrtcbin)?;
         add_audio_source(&pipeline, &webrtcbin)?;
diff --git a/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs b/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs
index 15f79db8ed..f7de6a912e 100644
--- a/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs
+++ b/webrtc/sendrecv/gst-sharp/WebRTCSendRecv.cs
@@ -16,10 +16,10 @@ namespace GstWebRTCDemo
     {
         const string SERVER = "wss://127.0.0.1:8443";
 
-        const string PIPELINE_DESC = @"webrtcbin name=sendrecv
- videotestsrc pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
+        const string PIPELINE_DESC = @"webrtcbin name=sendrecv bundle-policy=max-bundle
+ videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
  queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
- audiotestsrc wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+ audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
  queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.";
 
         readonly int _id;
@@ -306,4 +306,4 @@ namespace GstWebRTCDemo
         }
     }
 
-}
\ No newline at end of file
+}
diff --git a/webrtc/sendrecv/gst/webrtc-sendrecv.c b/webrtc/sendrecv/gst/webrtc-sendrecv.c
index b6bc6ff762..685ab79478 100644
--- a/webrtc/sendrecv/gst/webrtc-sendrecv.c
+++ b/webrtc/sendrecv/gst/webrtc-sendrecv.c
@@ -330,10 +330,10 @@ start_pipeline (void)
   GError *error = NULL;
 
   pipe1 =
-      gst_parse_launch ("webrtcbin name=sendrecv " STUN_SERVER
-      "videotestsrc pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
+      gst_parse_launch ("webrtcbin bundle-policy=max-bundle name=sendrecv " STUN_SERVER
+      "videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay ! "
       "queue ! " RTP_CAPS_VP8 "96 ! sendrecv. "
-      "audiotestsrc wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
+      "audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay ! "
       "queue ! " RTP_CAPS_OPUS "97 ! sendrecv. ",
       &error);
 
diff --git a/webrtc/sendrecv/gst/webrtc-sendrecv.py b/webrtc/sendrecv/gst/webrtc-sendrecv.py
index e5220c25ab..88870b12ca 100644
--- a/webrtc/sendrecv/gst/webrtc-sendrecv.py
+++ b/webrtc/sendrecv/gst/webrtc-sendrecv.py
@@ -16,10 +16,10 @@ gi.require_version('GstSdp', '1.0')
 from gi.repository import GstSdp
 
 PIPELINE_DESC = '''
-webrtcbin name=sendrecv
- videotestsrc pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
+webrtcbin name=sendrecv bundle-policy=max-bundle
+ videotestsrc is-live=true pattern=ball ! videoconvert ! queue ! vp8enc deadline=1 ! rtpvp8pay !
  queue ! application/x-rtp,media=video,encoding-name=VP8,payload=97 ! sendrecv.
- audiotestsrc wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
+ audiotestsrc is-live=true wave=red-noise ! audioconvert ! audioresample ! queue ! opusenc ! rtpopuspay !
  queue ! application/x-rtp,media=audio,encoding-name=OPUS,payload=96 ! sendrecv.
 '''