Clear FrameBuffer if there were no frames received for 10 minutes

This is a workaround for the case when there are no video frames in a
call for a very long time, such that RTP timestamps wraparound and
FrameBuffer can't figure out if the frame is older or newer.

Bug: webrtc:9974
Change-Id: Ie1eaa4938813dbbd637ddcbe7ff118ead2bfa4a9
Reviewed-on: https://webrtc-review.googlesource.com/c/109882
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#25548}

modules/video_coding/frame_buffer2.cc

@@ -113,6 +113,9 @@ FrameBuffer::ReturnReason FrameBuffer::NextFrame(
         if (keyframe_required && !frame->is_keyframe())
           continue;
 
+        // TODO(https://bugs.webrtc.org/9974): consider removing this check
+        // as it may make a stream undecodable after a very long delay between
+        // frames.
         if (last_decoded_frame_timestamp_ &&
             AheadOf(*last_decoded_frame_timestamp_, frame->Timestamp())) {
           continue;
@@ -244,6 +247,11 @@ void FrameBuffer::Stop() {
   new_continuous_frame_event_.Set();
 }
 
+void FrameBuffer::Clear() {
+  rtc::CritScope lock(&crit_);
+  ClearFramesAndHistory();
+}
+
 void FrameBuffer::UpdateRtt(int64_t rtt_ms) {
   rtc::CritScope lock(&crit_);
   jitter_estimator_->UpdateRtt(rtt_ms);

modules/video_coding/frame_buffer2.h

@@ -79,6 +79,9 @@ class FrameBuffer {
   // Updates the RTT for jitter buffer estimation.
   void UpdateRtt(int64_t rtt_ms);
 
+  // Clears the FrameBuffer, removing all the buffered frames.
+  void Clear();
+
  private:
   struct FrameInfo {
     FrameInfo();

video/video_receive_stream.cc

@@ -110,6 +110,11 @@ class NullVideoDecoder : public webrtc::VideoDecoder {
   const char* ImplementationName() const override { return "NullVideoDecoder"; }
 };
 
+// TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
+// Maximum time between frames before resetting the FrameBuffer to avoid RTP
+// timestamps wraparound to affect FrameBuffer.
+constexpr int kInactiveStreamThresholdMs = 600000;  //  10 minutes.
+
 }  // namespace
 
 namespace internal {
@@ -373,6 +378,14 @@ void VideoReceiveStream::OnCompleteFrame(
   // partially enabled inter-layer prediction.
   frame->id.spatial_layer = 0;
 
+  // TODO(https://bugs.webrtc.org/9974): Consider removing this workaround.
+  int64_t time_now_ms = rtc::TimeMillis();
+  if (last_complete_frame_time_ms_ > 0 &&
+      time_now_ms - last_complete_frame_time_ms_ > kInactiveStreamThresholdMs) {
+    frame_buffer_->Clear();
+  }
+  last_complete_frame_time_ms_ = time_now_ms;
+
   int64_t last_continuous_pid = frame_buffer_->InsertFrame(std::move(frame));
   if (last_continuous_pid != -1)
     rtp_video_stream_receiver_.FrameContinuous(last_continuous_pid);

video/video_receive_stream.h

@@ -146,6 +146,7 @@ class VideoReceiveStream : public webrtc::VideoReceiveStream,
   bool frame_decoded_ = false;
 
   int64_t last_keyframe_request_ms_ = 0;
+  int64_t last_complete_frame_time_ms_ = 0;
 };
 }  // namespace internal
 }  // namespace webrtc