Fix stats for encoder target bitrate when target rate is zero.

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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 <algorithm>
@@ skipping 2455 matching lines @@
         return 0;
       rtc::CritScope lock(&crit_);
       bitrate_kbps_ = new_target_bitrate;
       observation_complete_.Set();
       return 0;
     }
 
     void PerformTest() override {
       ASSERT_TRUE(Wait())
           << "Timed out while waiting for encoder SetRates() call.";
-      // Wait for GetStats to report a corresponding bitrate.
+      WaitForEncoderTargetBitrateMatchStats();
+      send_stream_->Stop();
+      WaitForStatsReportZeroTargetBitrate();
+      send_stream_->Start();
+      WaitForEncoderTargetBitrateMatchStats();
+    }
+
+    void WaitForEncoderTargetBitrateMatchStats() {
       for (int i = 0; i < kDefaultTimeoutMs; ++i) {
         VideoSendStream::Stats stats = send_stream_->GetStats();
         {
           rtc::CritScope lock(&crit_);
           if ((stats.target_media_bitrate_bps + 500) / 1000 ==
               static_cast<int>(bitrate_kbps_)) {
             return;
           }
         }
         SleepMs(1);
       }
       FAIL()
           << "Timed out waiting for stats reporting the currently set bitrate.";
     }
 
+    void WaitForStatsReportZeroTargetBitrate() {
+      for (int i = 0; i < kDefaultTimeoutMs; ++i) {
+        if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
+          return;
+        }
+        SleepMs(1);
+      }
+      FAIL() << "Timed out waiting for stats reporting zero bitrate.";
+    }
+
    private:
     rtc::CriticalSection crit_;
     VideoSendStream* send_stream_;
     uint32_t bitrate_kbps_ GUARDED_BY(crit_);
   } test;
 
   RunBaseTest(&test);
 }
 
 TEST_F(EndToEndTest, GetStats) {
@@ skipping 1056 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc