fix: invalid stats panic

Author: theomonnom

libwebrtc/src/native/peer_connection.rs

@@ -452,6 +452,11 @@ impl PeerConnection {
                 .downcast::<oneshot::Sender<Result<Vec<RtcStats>, RtcError>>>()
                 .unwrap();
 
+            if stats.is_empty() {
+                let _ = tx.send(Ok(vec![]));
+                return;
+            }
+
             // Unwrap because it should not happens
             let vec = serde_json::from_str(&stats).unwrap();
             let _ = tx.send(Ok(vec));

libwebrtc/src/native/rtp_receiver.rs

@@ -46,6 +46,11 @@ impl RtpReceiver {
                 .downcast::<oneshot::Sender<Result<Vec<RtcStats>, RtcError>>>()
                 .unwrap();
 
+            if stats.is_empty() {
+                let _ = tx.send(Ok(vec![]));
+                return;
+            }
+
             // Unwrap because it should not happens
             let vec = serde_json::from_str(&stats).unwrap();
             let _ = tx.send(Ok(vec));

libwebrtc/src/native/rtp_sender.rs

@@ -46,6 +46,11 @@ impl RtpSender {
                 .downcast::<oneshot::Sender<Result<Vec<RtcStats>, RtcError>>>()
                 .unwrap();
 
+            if stats.is_empty() {
+                let _ = tx.send(Ok(vec![]));
+                return;
+            }
+
             // Unwrap because it should not happens
             let vec = serde_json::from_str(&stats).unwrap();
             let _ = tx.send(Ok(vec));

livekit/src/room/track/local_audio_track.rs

@@ -121,7 +121,7 @@ impl LocalAudioTrack {
     }
 
     pub async fn get_stats(&self) -> RoomResult<Vec<RtcStats>> {
-        super::get_stats(&self.inner).await
+        super::local_track::get_stats(&self.inner).await
     }
 
     pub(crate) fn on_muted(&self, f: impl Fn(Track) + Send + 'static) {

livekit/src/room/track/local_track.rs

@@ -12,9 +12,12 @@
 // See the License for the specific language governing permissions and
 // limitations under the License.
 
-use super::track_dispatch;
+use std::sync::Arc;
+
+use super::{track_dispatch, TrackInner};
 use crate::prelude::*;
 use libwebrtc::prelude::*;
+use libwebrtc::stats::RtcStats;
 use livekit_protocol as proto;
 use livekit_protocol::enum_dispatch;
 
@@ -61,3 +64,12 @@ impl TryFrom<Track> for LocalTrack {
         }
     }
 }
+
+pub(super) async fn get_stats(inner: &Arc<TrackInner>) -> RoomResult<Vec<RtcStats>> {
+    let transceiver = inner.info.read().transceiver.clone();
+    let Some(transceiver) = transceiver.as_ref() else {
+        return Err(RoomError::Internal("no transceiver found for track".into()));
+    };
+
+    Ok(transceiver.sender().get_stats().await?)
+}

livekit/src/room/track/local_video_track.rs

@@ -121,7 +121,7 @@ impl LocalVideoTrack {
     }
 
     pub async fn get_stats(&self) -> RoomResult<Vec<RtcStats>> {
-        super::get_stats(&self.inner).await
+        super::local_track::get_stats(&self.inner).await
     }
 
     pub(crate) fn on_muted(&self, f: impl Fn(Track) + Send + 'static) {

livekit/src/room/track/mod.rs

@@ -173,16 +173,6 @@ pub(super) fn new_inner(
     }
 }
 
-pub(super) async fn get_stats(inner: &Arc<TrackInner>) -> RoomResult<Vec<RtcStats>> {
-    let transceiver = inner.info.read().transceiver.clone();
-    let Some(transceiver) = transceiver.as_ref() else {
-        return Err(RoomError::Internal("no transceiver found for track".into()));
-    };
-
-    let rtp_receiver = transceiver.receiver();
-    Ok(rtp_receiver.get_stats().await?)
-}
-
 /// This is only called for local tracks
 pub(super) fn set_muted(inner: &Arc<TrackInner>, track: &Track, muted: bool) {
     let info = inner.info.read();

livekit/src/room/track/remote_audio_track.rs

@@ -91,7 +91,7 @@ impl RemoteAudioTrack {
     }
 
     pub async fn get_stats(&self) -> RoomResult<Vec<RtcStats>> {
-        super::get_stats(&self.inner).await
+        super::remote_track::get_stats(&self.inner).await
     }
 
     pub(crate) fn on_muted(&self, f: impl Fn(Track) + Send + 'static) {

livekit/src/room/track/remote_track.rs

@@ -16,7 +16,6 @@ use super::track_dispatch;
 use super::TrackInner;
 use crate::prelude::*;
 use libwebrtc::prelude::*;
-use libwebrtc::stats::InboundRtpStats;
 use libwebrtc::stats::RtcStats;
 use livekit_protocol as proto;
 use livekit_protocol::enum_dispatch;
@@ -40,6 +39,15 @@ impl RemoteTrack {
     }
 }
 
+pub(super) async fn get_stats(inner: &Arc<TrackInner>) -> RoomResult<Vec<RtcStats>> {
+    let transceiver = inner.info.read().transceiver.clone();
+    let Some(transceiver) = transceiver.as_ref() else {
+        return Err(RoomError::Internal("no transceiver found for track".into()));
+    };
+
+    Ok(transceiver.receiver().get_stats().await?)
+}
+
 pub(super) fn update_info(inner: &Arc<TrackInner>, track: &Track, new_info: proto::TrackInfo) {
     super::update_info(inner, track, new_info.clone());
     super::set_muted(inner, track, new_info.muted);

livekit/src/room/track/remote_video_track.rs

@@ -92,7 +92,7 @@ impl RemoteVideoTrack {
     }
 
     pub async fn get_stats(&self) -> RoomResult<Vec<RtcStats>> {
-        super::get_stats(&self.inner).await
+        super::remote_track::get_stats(&self.inner).await
     }
 
     pub(crate) fn on_muted(&self, f: impl Fn(Track) + Send + 'static) {