More gracefully handle timing errors, such as unexpected changes in the rtp timestamp.

webrtc/modules/video_coding/frame_buffer2.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 123 matching lines @@
         if (inter_frame_delay_.CalculateDelay(frame->timestamp, &frame_delay,
                                               frame->ReceivedTime())) {
           jitter_estimator_->UpdateEstimate(frame_delay, frame->size());
         }
 
         float rtt_mult = protection_mode_ == kProtectionNackFEC ? 0.0 : 1.0;
         timing_->SetJitterDelay(jitter_estimator_->GetJitterEstimate(rtt_mult));
         timing_->UpdateCurrentDelay(frame->RenderTime(), now_ms);
       }
 
+      // Gracefully handle bad RTP timestamps and render time issues.
+      if (HasBadRenderTiming(*frame, now_ms)) {
+        jitter_estimator_->Reset();
+        timing_->Reset();
+        frame->SetRenderTime(timing_->RenderTimeMs(frame->timestamp, now_ms));
+      }
+
       UpdateJitterDelay();
       PropagateDecodability(next_frame_it_->second);
 
       // Sanity check for RTP timestamp monotonicity.
       if (last_decoded_frame_it_ != frames_.end()) {
         const FrameKey& last_decoded_frame_key = last_decoded_frame_it_->first;
         const FrameKey& frame_key = next_frame_it_->first;
 
         const bool frame_is_higher_spatial_layer_of_last_decoded_frame =
             last_decoded_frame_timestamp_ == frame->timestamp &&
@@ skipping 28 matching lines @@
     // If |next_frame_it_ == frames_.end()| and there is still time left, it
     // means that the frame buffer was cleared as the thread in this function
     // was waiting to acquire |crit_| in order to return. Wait for the
     // remaining time and then return.
     return NextFrame(latest_return_time_ms - now_ms, frame_out);
   }
 
   return kTimeout;
 }
 
+bool FrameBuffer::HasBadRenderTiming(const FrameObject& frame, int64_t now_ms) {
+  // Assume that render timing errors are due to changes in the video stream.
+  int64_t render_time_ms = frame.RenderTimeMs();
+  const int64_t kMaxVideoDelayMs = 10000;
+  if (render_time_ms < 0) {
+    return true;
+  }
+  if (std::abs(render_time_ms - now_ms) > kMaxVideoDelayMs) {
+    int frame_delay = static_cast<int>(std::abs(render_time_ms - now_ms));
+    LOG(LS_WARNING) << "A frame about to be decoded is out of the configured "
+                    << "delay bounds (" << frame_delay << " > "
+                    << kMaxVideoDelayMs
+                    << "). Resetting the video jitter buffer.";
+    return true;
+  }
+  if (static_cast<int>(timing_->TargetVideoDelay()) > kMaxVideoDelayMs) {
+    LOG(LS_WARNING) << "The video target delay has grown larger than "
+                    << kMaxVideoDelayMs << " ms.";
+    return true;
+  }
+  return false;
+}
+
 void FrameBuffer::SetProtectionMode(VCMVideoProtection mode) {
   TRACE_EVENT0("webrtc", "FrameBuffer::SetProtectionMode");
   rtc::CritScope lock(&crit_);
   protection_mode_ = mode;
 }
 
 void FrameBuffer::Start() {
   TRACE_EVENT0("webrtc", "FrameBuffer::Start");
   rtc::CritScope lock(&crit_);
   stopped_ = false;
@@ skipping 287 matching lines @@
   frames_.clear();
   last_decoded_frame_it_ = frames_.end();
   last_continuous_frame_it_ = frames_.end();
   next_frame_it_ = frames_.end();
   num_frames_history_ = 0;
   num_frames_buffered_ = 0;
 }
 
 }  // namespace video_coding
 }  // namespace webrtc