From: Sebastian Dröge <sebastian@centricular.com>
Date: Thu, 19 Jan 2023 18:38:41 +0000 (+0200)
Subject: examples: webrtc: rust: Set keyframe-max-dist=2000 and picture-id-mode=15-bit for... 
X-Git-Tag: 1.22.0~14
X-Git-Url: http://review.tizen.org/git/?a=commitdiff_plain;h=6541dccaeac0abf2dfc64ab92ff1706780bfe2bc;p=platform%2Fupstream%2Fgstreamer.git

examples: webrtc: rust: Set keyframe-max-dist=2000 and picture-id-mode=15-bit for VP8 and perfect-timestamps=true for audio

This makes it in sync with the C sendrecv and generally behaves better.

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/3758>
---

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 b00cda9..8a9aed2 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
@@ -157,9 +157,9 @@ impl App {
         // Create the GStreamer pipeline
         let pipeline = gst::parse_launch(
             &format!(
-                "videotestsrc is-live=true ! vp8enc deadline=1 ! rtpvp8pay pt=96 ! tee name=video-tee ! \
+                "videotestsrc is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay pt=96 picture-id-mode=15-bit ! tee name=video-tee ! \
                  queue ! fakesink sync=true \
-                 audiotestsrc wave=ticks is-live=true ! opusenc ! rtpopuspay pt=97 ! application/x-rtp,encoding-name=OPUS ! tee name=audio-tee ! \
+                 audiotestsrc wave=ticks is-live=true ! opusenc perfect-timestamp=true ! 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 \
@@ -192,13 +192,6 @@ impl App {
         // Channel for outgoing WebSocket messages from other threads
         let (send_ws_msg_tx, send_ws_msg_rx) = mpsc::unbounded::<WsMessage>();
 
-        // Asynchronously set the pipeline to Playing
-        pipeline.call_async(|pipeline| {
-            pipeline
-                .set_state(gst::State::Playing)
-                .expect("Couldn't set pipeline to Playing");
-        });
-
         let app = App(Arc::new(AppInner {
             pipeline,
             video_tee,
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 ef613cd..7a53608 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 name=vpay pt=96 ! webrtcbin. \
+        "videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 keyframe-max-dist=2000 ! rtpvp8pay name=vpay pt=96 picture-id-mode=15-bit ! webrtcbin. \
          audiotestsrc is-live=true ! opusenc perfect-timestamp=true ! rtpopuspay name=apay pt=97 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \
          webrtcbin name=webrtcbin"
     )?;