Store first_frame as const& instead of *

Bug: webrtc:13343
Change-Id: Id6d73539fa3034be9e7d4e6a27ca5b615ad204da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/236842
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#35291}
diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index bbbe76d..b7ae0f3 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -249,14 +249,14 @@
   RTC_DCHECK(!frames_to_decode_.empty());
   bool superframe_delayed_by_retransmission = false;
   size_t superframe_size = 0;
-  EncodedFrame* first_frame = frames_to_decode_[0]->second.frame.get();
-  int64_t render_time_ms = first_frame->RenderTime();
-  int64_t receive_time_ms = first_frame->ReceivedTime();
+  const EncodedFrame& first_frame = *frames_to_decode_[0]->second.frame;
+  int64_t render_time_ms = first_frame.RenderTime();
+  int64_t receive_time_ms = first_frame.ReceivedTime();
   // Gracefully handle bad RTP timestamps and render time issues.
-  if (HasBadRenderTiming(*first_frame, now_ms)) {
+  if (HasBadRenderTiming(first_frame, now_ms)) {
     jitter_estimator_.Reset();
     timing_->Reset();
-    render_time_ms = timing_->RenderTimeMs(first_frame->Timestamp(), now_ms);
+    render_time_ms = timing_->RenderTimeMs(first_frame.Timestamp(), now_ms);
   }
 
   for (FrameMap::iterator& frame_it : frames_to_decode_) {
@@ -292,8 +292,8 @@
   if (!superframe_delayed_by_retransmission) {
     int64_t frame_delay;
 
-    if (inter_frame_delay_.CalculateDelay(first_frame->Timestamp(),
-                                          &frame_delay, receive_time_ms)) {
+    if (inter_frame_delay_.CalculateDelay(first_frame.Timestamp(), &frame_delay,
+                                          receive_time_ms)) {
       jitter_estimator_.UpdateEstimate(frame_delay, superframe_size);
     }