Replace voe_conference_test.

webrtc/pc/peerconnection_integrationtest.cc

 /*
  *  Copyright 2012 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 1896 matching lines @@
       expected_caller_received_frames, expected_caller_received_frames,
       kDefaultExpectedAudioFrameCount, kDefaultExpectedVideoFrameCount,
       kMaxWaitForFramesMs);
 
   // The callee received frames, so we definitely should have nonzero "sent
   // bytes" stats at this point.
   EXPECT_GT(caller()->OldGetStatsForTrack(audio_track)->BytesSent(), 0);
   EXPECT_GT(caller()->OldGetStatsForTrack(video_track)->BytesSent(), 0);
 }
 
+// Test that we can get capture start ntp time.
+TEST_F(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddAudioOnlyMediaStream();
+
+  auto audio_track = callee()->CreateLocalAudioTrack();
+  callee()->AddMediaStreamFromTracks(audio_track, nullptr);
+
+  // Do offer/answer, wait for the callee to receive some frames.
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+  // Get the remote audio track created on the receiver, so they can be used as
+  // GetStats filters.
+  StreamCollectionInterface* remote_streams = callee()->remote_streams();
+  ASSERT_EQ(1u, remote_streams->count());
+  ASSERT_EQ(1u, remote_streams->at(0)->GetAudioTracks().size());
+  MediaStreamTrackInterface* remote_audio_track =
+      remote_streams->at(0)->GetAudioTracks()[0];
+
+  // Get the audio output level stats. Note that the level is not available
+  // until an RTCP packet has been received.
+  EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track)->
+                   CaptureStartNtpTime() > 0, 2 * kMaxWaitForFramesMs);
+}
+
 // Test that we can get stats (using the new stats implemnetation) for
 // unsignaled streams. Meaning when SSRCs/MSIDs aren't signaled explicitly in
 // SDP.
 TEST_F(PeerConnectionIntegrationTest,
        GetStatsForUnsignaledStreamWithNewStatsApi) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioOnlyMediaStream();
   // Remove SSRCs and MSIDs from the received offer SDP.
   callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids);
@@ skipping 1044 matching lines @@
   caller()->CreateAndSetAndSignalOffer();
   ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
   // Wait for additional audio frames to be received by the callee.
   ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
                                   kMaxWaitForFramesMs);
 }
 
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)