Remove use-after-free when quality tests stall.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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.
  */
 #include <stdio.h>
@@ skipping 55 matching lines @@
         frames_to_process_(duration_frames),
         frames_recorded_(0),
         frames_processed_(0),
         dropped_frames_(0),
         last_render_time_(0),
         rtp_timestamp_delta_(0),
         avg_psnr_threshold_(avg_psnr_threshold),
         avg_ssim_threshold_(avg_ssim_threshold),
         stats_polling_thread_(&PollStatsThread, this, "StatsPoller"),
         comparison_available_event_(false, false),
-        done_(false, false) {
+        done_(true, false) {
     // Create thread pool for CPU-expensive PSNR/SSIM calculations.
 
     // Try to use about as many threads as cores, but leave kMinCoresLeft alone,
     // so that we don't accidentally starve "real" worker threads (codec etc).
     // Also, don't allocate more than kMaxComparisonThreads, even if there are
     // spare cores.
 
     uint32_t num_cores = CpuInfo::DetectNumberOfCores();
     RTC_DCHECK_GE(num_cores, 1u);
     static const uint32_t kMinCoresLeft = 4;
@@ skipping 149 matching lines @@
       const char* kKeepAliveMessages[3] = {
           "Uh, I'm-I'm not quite dead, sir.",
           "Uh, I-I think uh, I could pull through, sir.",
           "Actually, I think I'm all right to come with you--"};
       printf("- %s\n", kKeepAliveMessages[iteration++ % 3]);
 
       if (last_frames_processed == -1) {
         last_frames_processed = frames_processed;
         continue;
       }
-      ASSERT_GT(frames_processed, last_frames_processed)

[pbos-webrtc, 2016/01/19 13:37:28]

This ASSERT caused the function to abort before stats_polling_thread_.Stop()
could trigger.

-          << "Analyzer stalled while waiting for test to finish.";
+      if (frames_processed == last_frames_processed) {
+        EXPECT_GT(frames_processed, last_frames_processed)
+            << "Analyzer stalled while waiting for test to finish.";
+        done_.Set();
+        break;
+      }
       last_frames_processed = frames_processed;
     }
 
     if (iteration > 0)
       printf("- Farewell, sweet Concorde!\n");
 
-    // Signal stats polling thread if that is still waiting and stop it now,
-    // since it uses the send_stream_ reference that might be reclaimed after
-    // returning from this method.
-    done_.Set();
     stats_polling_thread_.Stop();
   }
 
   VideoCaptureInput* input_;
   test::LayerFilteringTransport* const transport_;
   PacketReceiver* receiver_;
   VideoSendStream* send_stream_;
 
  private:
   struct FrameComparison {
@@ skipping 84 matching lines @@
                                            send_time_ms, recv_time_ms,
                                            render_time_ms, encoded_size));
     comparison_available_event_.Set();
   }
 
   static bool PollStatsThread(void* obj) {
     return static_cast<VideoAnalyzer*>(obj)->PollStats();
   }
 
   bool PollStats() {
-    if (done_.Wait(kSendStatsPollingIntervalMs)) {
-      // Set event again to make sure main thread is also signaled, then we're
-      // done.
-      done_.Set();
+    if (done_.Wait(kSendStatsPollingIntervalMs))
       return false;
-    }
 
     VideoSendStream::Stats stats = send_stream_->GetStats();
 
     rtc::CritScope crit(&comparison_lock_);
     encode_frame_rate_.AddSample(stats.encode_frame_rate);
     encode_time_ms.AddSample(stats.avg_encode_time_ms);
     encode_usage_percent.AddSample(stats.encode_usage_percent);
     media_bitrate_bps.AddSample(stats.media_bitrate_bps);
 
     return true;
@@ skipping 686 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc