- Removes voe_conference_test.
- Adds a new AudioStatsTest, with better coverage of the same features, based on call_test.
- Adds an AudioEndToEndTest utility, which AudioStatsTest and LowBandwidthAudioTest uses.
BUG=webrtc:4690
R=kwiberg@webrtc.org
Review-Url: https://codereview.webrtc.org/3008273002 .
Cr-Commit-Position: refs/heads/master@{#19833}
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc
index 641be6f..6760b14 100644
--- a/webrtc/pc/peerconnection_integrationtest.cc
+++ b/webrtc/pc/peerconnection_integrationtest.cc
@@ -1914,6 +1914,33 @@
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.
