examples: webrtc: sendrecv: rust: Use the correct payload types if the remote is...

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

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 c06d777..ef613cd 100644 (file)
--- a/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
+++ b/subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs
@@ -115,8 +115,8 @@ 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 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \
+        "videotestsrc pattern=ball is-live=true ! vp8enc deadline=1 ! rtpvp8pay name=vpay pt=96 ! webrtcbin. \
+         audiotestsrc is-live=true ! opusenc perfect-timestamp=true ! rtpopuspay name=apay pt=97 ! application/x-rtp,encoding-name=OPUS ! webrtcbin. \
          webrtcbin name=webrtcbin"
     )?;
 
@@ -200,24 +200,20 @@ impl App {
             }
         });
 
-        // Asynchronously set the pipeline to Playing
-        app.pipeline.call_async(|pipeline| {
-            // If this fails, post an error on the bus so we exit
-            if pipeline.set_state(gst::State::Playing).is_err() {
-                gst::element_error!(
-                    pipeline,
-                    gst::LibraryError::Failed,
-                    ("Failed to set pipeline to Playing")
-                );
-            }
-        });
-
-        // Asynchronously set the pipeline to Playing
-        app.pipeline.call_async(|pipeline| {
-            pipeline
-                .set_state(gst::State::Playing)
-                .expect("Couldn't set pipeline to Playing");
-        });
+        // Asynchronously set the pipeline to Playing if we're creating the offer,
+        // otherwise do that after the offer was received.
+        if app.args.peer_id.is_some() {
+            app.pipeline.call_async(|pipeline| {
+                // If this fails, post an error on the bus so we exit
+                if pipeline.set_state(gst::State::Playing).is_err() {
+                    gst::element_error!(
+                        pipeline,
+                        gst::LibraryError::Failed,
+                        ("Failed to set pipeline to Playing")
+                    );
+                }
+            });
+        }
 
         Ok((app, send_gst_msg_rx, send_ws_msg_rx))
     }
@@ -377,6 +373,63 @@ impl App {
         Ok(())
     }
 
+    fn configure_pipeline_on_offer(
+        &self,
+        offer: &gst_sdp::SDPMessage,
+    ) -> Result<(), anyhow::Error> {
+        // Extract audio/video payload types from the SDP and configure accordingly on the
+        // pipeline as these have to match with the offer
+        let mut opus_id = None;
+        let mut vp8_id = None;
+        for media in offer.medias() {
+            for fmt in media.formats() {
+                if fmt == "webrtc-datachannel" {
+                    continue;
+                }
+
+                let pt = match fmt.parse::<u8>() {
+                    Ok(pt) => pt,
+                    Err(_) => continue,
+                };
+
+                let caps = match media.caps_from_media(pt as i32) {
+                    Some(caps) if caps.size() > 0 => caps,
+                    _ => continue,
+                };
+
+                let s = caps.structure(0).unwrap();
+                let encoding_name = match s.get::<&str>("encoding-name") {
+                    Ok(encoding_name) => encoding_name,
+                    Err(_) => continue,
+                };
+
+                if encoding_name == "VP8" && vp8_id.is_none() {
+                    vp8_id = Some(pt);
+                } else if encoding_name == "OPUS" && opus_id.is_none() {
+                    opus_id = Some(pt);
+                }
+            }
+        }
+
+        if let (Some(opus_id), Some(vp8_id)) = (opus_id, vp8_id) {
+            let apay = self.pipeline.by_name("apay").unwrap();
+            let vpay = self.pipeline.by_name("vpay").unwrap();
+
+            for (pay, pt) in [(apay, opus_id), (vpay, vp8_id)] {
+                pay.set_property("pt", pt as u32);
+            }
+        } else {
+            gst::element_error!(
+                self.pipeline,
+                gst::LibraryError::Failed,
+                ("Not all streams found in the offer")
+            );
+            bail!("Not all streams found in the offer");
+        }
+
+        Ok(())
+    }
+
     // Handle incoming SDP answers from the peer
     fn handle_sdp(&self, type_: &str, sdp: &str) -> Result<(), anyhow::Error> {
         if type_ == "answer" {
@@ -403,6 +456,20 @@ impl App {
             self.pipeline.call_async(move |_pipeline| {
                 let app = upgrade_weak!(app_clone);
 
+                if app.configure_pipeline_on_offer(&ret).is_err() {
+                    return;
+                }
+
+                // If this fails, post an error on the bus so we exit
+                if app.pipeline.set_state(gst::State::Playing).is_err() {
+                    gst::element_error!(
+                        app.pipeline,
+                        gst::LibraryError::Failed,
+                        ("Failed to set pipeline to Playing")
+                    );
+                    return;
+                }
+
                 let offer = gst_webrtc::WebRTCSessionDescription::new(
                     gst_webrtc::WebRTCSDPType::Offer,
                     ret,