examples: webrtc: rust: Update to gstreamer-rs 0.21

Part-of: <https://gitlab.freedesktop.org/gstreamer/gstreamer/-/merge_requests/5181>

subprojects/gst-examples/webrtc/janus/rust/Cargo.toml

@@ -11,15 +11,15 @@ clap = { version = "4", features = ["derive"] }
 anyhow = "1"
 url = "2"
 rand = "0.8"
-async-tungstenite = { version = "0.20", features = ["gio-runtime"] }
-gst = { package = "gstreamer", version = "0.20" }
-gst-webrtc = { package = "gstreamer-webrtc", version = "0.20" }
-gst-sdp = { package = "gstreamer-sdp", version = "0.20" }
+async-tungstenite = { version = "0.23", features = ["gio-runtime"] }
+gst = { package = "gstreamer", version = "0.21" }
+gst-webrtc = { package = "gstreamer-webrtc", version = "0.21" }
+gst-sdp = { package = "gstreamer-sdp", version = "0.21" }
 serde = "1"
 serde_derive = "1"
 serde_json = "1.0.53"
 http = "0.2"
-glib = "0.17"
-gio = "0.17"
+glib = "0.18"
+gio = "0.18"
 log = "0.4.8"
 env_logger = "0.10"

subprojects/gst-examples/webrtc/janus/rust/src/janus.rs

@@ -503,10 +503,11 @@ impl JanusGateway {
             .static_pad("sink")
             .expect("Failed to get sink pad from encoder");
 
-        if let Ok(video_ghost_pad) = gst::GhostPad::with_target(Some("video_sink"), &sinkpad) {
-            encode_bin.add_pad(&video_ghost_pad)?;
-            srcpad.link(&video_ghost_pad)?;
-        }
+        let video_ghost_pad = gst::GhostPad::builder_with_target(&sinkpad)?
+            .name("video_sink")
+            .build();
+        encode_bin.add_pad(&video_ghost_pad)?;
+        srcpad.link(&video_ghost_pad)?;
 
         let sinkpad2 = webrtcbin
             .request_pad_simple("sink_%u")
@@ -515,11 +516,11 @@ impl JanusGateway {
             .by_name("webrtc-vsink")
             .expect("No webrtc-vsink found");
         let srcpad = vsink.static_pad("src").expect("Element without src pad");
-        if let Ok(webrtc_ghost_pad) = gst::GhostPad::with_target(Some("webrtc_video_src"), &srcpad)
-        {
-            encode_bin.add_pad(&webrtc_ghost_pad)?;
-            webrtc_ghost_pad.link(&sinkpad2)?;
-        }
+        let webrtc_ghost_pad = gst::GhostPad::builder_with_target(&srcpad)?
+            .name("webrtc_video_src")
+            .build();
+        encode_bin.add_pad(&webrtc_ghost_pad)?;
+        webrtc_ghost_pad.link(&sinkpad2)?;
 
         let transceiver = webrtcbin.emit_by_name::<glib::Object>("get-transceiver", &[&0i32]);
         transceiver.set_property("do-nack", false);

subprojects/gst-examples/webrtc/janus/rust/src/main.rs

@@ -20,7 +20,7 @@
 
 use anyhow::bail;
 use gst::prelude::*;
-use std::sync::{Arc, Weak};
+use std::sync::{Arc, Mutex, Weak};
 
 mod janus;
 
@@ -36,6 +36,7 @@ struct AppWeak(Weak<AppInner>);
 #[derive(Debug)]
 struct AppInner {
     pipeline: gst::Pipeline,
+    bus_watch: Mutex<Option<gst::bus::BusWatchGuard>>,
 }
 
 // To be able to access the App's fields directly
@@ -71,18 +72,24 @@ impl App {
             .expect("Couldn't downcast pipeline");
 
         let bus = pipeline.bus().unwrap();
-        let app = App(Arc::new(AppInner { pipeline }));
+        let app = App(Arc::new(AppInner {
+            pipeline,
+            bus_watch: Mutex::default(),
+        }));
 
         let app_weak = app.downgrade();
-        bus.add_watch_local(move |_bus, msg| {
-            let app = upgrade_weak!(app_weak, glib::Continue(false));
+        let bus_watch = bus
+            .add_watch_local(move |_bus, msg| {
+                let app = upgrade_weak!(app_weak, glib::ControlFlow::Break);
+
+                if app.handle_pipeline_message(msg).is_err() {
+                    return glib::ControlFlow::Break;
+                }
+                glib::ControlFlow::Continue
+            })
+            .expect("Unable to add bus watch");
+        *app.0.bus_watch.lock().unwrap() = Some(bus_watch);
 
-            if app.handle_pipeline_message(msg).is_err() {
-                return glib::Continue(false);
-            }
-            glib::Continue(true)
-        })
-        .expect("Unable to add bus watch");
         Ok(app)
     }
 

subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/Cargo.toml

@@ -10,13 +10,13 @@ async-std = "1"
 clap = { version = "4", features = ["derive"] }
 anyhow = "1"
 rand = "0.8"
-async-tungstenite = { version = "0.20", features = ["async-std-runtime", "async-native-tls"] }
-gst = { package = "gstreamer", version = "0.20" }
-gst-webrtc = { package = "gstreamer-webrtc", version = "0.20" }
-gst-sdp = { package = "gstreamer-sdp", version = "0.20" }
+async-tungstenite = { version = "0.23", features = ["async-std-runtime", "async-native-tls"] }
+gst = { package = "gstreamer", version = "0.21" }
+gst-webrtc = { package = "gstreamer-webrtc", version = "0.21" }
+gst-sdp = { package = "gstreamer-sdp", version = "0.21" }
 serde = "1"
 serde_derive = "1"
 serde_json = "1"
 
 [target.'cfg(target_os = "macos")'.dependencies]
-cocoa = "0.24"
+cocoa = "0.25"

subprojects/gst-examples/webrtc/multiparty-sendrecv/gst-rust/src/main.rs

@@ -321,21 +321,21 @@ impl App {
         let audio_queue = peer_bin
             .by_name("audio-queue")
             .expect("can't find audio-queue");
-        let audio_sink_pad = gst::GhostPad::with_target(
-            Some("audio_sink"),
-            &audio_queue.static_pad("sink").unwrap(),
-        )
-        .unwrap();
+        let audio_sink_pad =
+            gst::GhostPad::builder_with_target(&audio_queue.static_pad("sink").unwrap())
+                .unwrap()
+                .name("audio_sink")
+                .build();
         peer_bin.add_pad(&audio_sink_pad).unwrap();
 
         let video_queue = peer_bin
             .by_name("video-queue")
             .expect("can't find video-queue");
-        let video_sink_pad = gst::GhostPad::with_target(
-            Some("video_sink"),
-            &video_queue.static_pad("sink").unwrap(),
-        )
-        .unwrap();
+        let video_sink_pad =
+            gst::GhostPad::builder_with_target(&video_queue.static_pad("sink").unwrap())
+                .unwrap()
+                .name("video_sink")
+                .build();
         peer_bin.add_pad(&video_sink_pad).unwrap();
 
         let peer = Peer(Arc::new(PeerInner {
@@ -837,14 +837,12 @@ impl Peer {
 
         // Add a ghost pad on our conv bin that proxies the sink pad of the decodebin
         let dbin = conv.by_name("dbin").unwrap();
-        let sinkpad =
-            gst::GhostPad::with_target(Some("sink"), &dbin.static_pad("sink").unwrap()).unwrap();
+        let sinkpad = gst::GhostPad::with_target(&dbin.static_pad("sink").unwrap()).unwrap();
         conv.add_pad(&sinkpad).unwrap();
 
         // And another one that proxies the source pad of the last element
         let src = conv.by_name("src").unwrap();
-        let srcpad =
-            gst::GhostPad::with_target(Some("src"), &src.static_pad("src").unwrap()).unwrap();
+        let srcpad = gst::GhostPad::with_target(&src.static_pad("src").unwrap()).unwrap();
         conv.add_pad(&srcpad).unwrap();
 
         self.bin.add(&conv).unwrap();
@@ -856,11 +854,17 @@ impl Peer {
 
         // And then add a new ghost pad to the peer bin that proxies the source pad we added above
         if media_type == "video" {
-            let srcpad = gst::GhostPad::with_target(Some("video_src"), &srcpad).unwrap();
+            let srcpad = gst::GhostPad::builder_with_target(&srcpad)
+                .unwrap()
+                .name("video_src")
+                .build();
             srcpad.set_active(true).unwrap();
             self.bin.add_pad(&srcpad).unwrap();
         } else if media_type == "audio" {
-            let srcpad = gst::GhostPad::with_target(Some("audio_src"), &srcpad).unwrap();
+            let srcpad = gst::GhostPad::builder_with_target(&srcpad)
+                .unwrap()
+                .name("audio_src")
+                .build();
             srcpad.set_active(true).unwrap();
             self.bin.add_pad(&srcpad).unwrap();
         }

subprojects/gst-examples/webrtc/sendrecv/gst-rust/Cargo.toml

@@ -10,14 +10,14 @@ async-std = "1"
 clap = { version = "4", features = ["derive"] }
 anyhow = "1"
 rand = "0.8"
-async-tungstenite = { version = "0.20", features = ["async-std-runtime", "async-native-tls"] }
-gst = { package = "gstreamer", version = "0.20" }
-gst-rtp = { package = "gstreamer-rtp", version = "0.20", features = ["v1_20"] }
-gst-webrtc = { package = "gstreamer-webrtc", version = "0.20" }
-gst-sdp = { package = "gstreamer-sdp", version = "0.20" }
+async-tungstenite = { version = "0.23", features = ["async-std-runtime", "async-native-tls"] }
+gst = { package = "gstreamer", version = "0.21" }
+gst-rtp = { package = "gstreamer-rtp", version = "0.21", features = ["v1_20"] }
+gst-webrtc = { package = "gstreamer-webrtc", version = "0.21" }
+gst-sdp = { package = "gstreamer-sdp", version = "0.21" }
 serde = "1"
 serde_derive = "1"
 serde_json = "1"
 
 [target.'cfg(target_os = "macos")'.dependencies]
-cocoa = "0.24"
+cocoa = "0.25"

subprojects/gst-examples/webrtc/sendrecv/gst-rust/src/main.rs

@@ -377,7 +377,7 @@ impl App {
                 bail!("Answer creation future got no response");
             }
             Err(err) => {
-                bail!("Answer creation future got error response: {err:?}", err);
+                bail!("Answer creation future got error response: {err:?}");
             }
         };