Change reporting of timing frames conditions in GetStats on receive side

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 3611 matching lines @@
     std::set<uint32_t> expected_send_ssrcs_;
     std::string expected_cname_;
 
     rtc::Event check_stats_event_;
     ReceiveStreamRenderer receive_stream_renderer_;
   } test;
 
   RunBaseTest(&test);
 }
 
-TEST_F(EndToEndTest, GetTimingFrameInfoReportsTimingFrames) {
+TEST_F(EndToEndTest, TimingFramesAreReported) {
   static const int kExtensionId = 5;
 
   class StatsObserver : public test::EndToEndTest {
    public:
     StatsObserver() : EndToEndTest(kLongTimeoutMs) {}
 
    private:
-    std::string CompoundKey(const char* name, uint32_t ssrc) {
-      std::ostringstream oss;
-      oss << name << "_" << ssrc;
-      return oss.str();
-    }
-
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.extensions.clear();
       send_config->rtp.extensions.push_back(
           RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
       for (size_t i = 0; i < receive_configs->size(); ++i) {
         (*receive_configs)[i].rtp.extensions.clear();
         (*receive_configs)[i].rtp.extensions.push_back(
             RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
       }
     }
 
     void OnVideoStreamsCreated(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       receive_streams_ = receive_streams;
     }
 
     void PerformTest() override {
       // No frames reported initially.
       for (size_t i = 0; i < receive_streams_.size(); ++i) {
-        EXPECT_FALSE(receive_streams_[i]->GetAndResetTimingFrameInfo());
+        EXPECT_FALSE(receive_streams_[i]->GetStats().timing_frame_info);
       }
       // Wait for at least one timing frame to be sent with 100ms grace period.
       SleepMs(kDefaultTimingFramesDelayMs + 100);
       // Check that timing frames are reported for each stream.
       for (size_t i = 0; i < receive_streams_.size(); ++i) {
-        EXPECT_TRUE(receive_streams_[i]->GetAndResetTimingFrameInfo());
+        EXPECT_TRUE(receive_streams_[i]->GetStats().timing_frame_info);
       }
     }
 
     std::vector<VideoReceiveStream*> receive_streams_;
   } test;
 
   RunBaseTest(&test);
 }
 
 class RtcpXrObserver : public test::EndToEndTest {
@@ skipping 1336 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc