Index: webrtc/pc/peerconnection_integrationtest.cc |
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc |
index ad1a12caf6d323ff3dbd4c93b4479896d2d85414..e6c3cf1f5781f18e493cbbb7f7d0cf3b83a6b6fc 100644 |
--- a/webrtc/pc/peerconnection_integrationtest.cc |
+++ b/webrtc/pc/peerconnection_integrationtest.cc |
@@ -116,6 +116,18 @@ void RemoveSsrcsAndMsids(cricket::SessionDescription* desc) { |
desc->set_msid_supported(false); |
} |
+int FindFirstMediaStatsIndexByKind( |
+ const std::string& kind, |
+ const std::vector<const webrtc::RTCMediaStreamTrackStats*>& |
+ media_stats_vec) { |
+ for (size_t i = 0; i < media_stats_vec.size(); i++) { |
+ if (media_stats_vec[i]->kind.ValueToString() == kind) { |
+ return i; |
+ } |
+ } |
+ return -1; |
+} |
+ |
class SignalingMessageReceiver { |
public: |
virtual void ReceiveSdpMessage(const std::string& type, |
@@ -1926,9 +1938,31 @@ TEST_F(PeerConnectionIntegrationTest, |
ASSERT_EQ(1U, inbound_stream_stats.size()); |
ASSERT_TRUE(inbound_stream_stats[0]->bytes_received.is_defined()); |
ASSERT_GT(*inbound_stream_stats[0]->bytes_received, 0U); |
- // TODO(deadbeef): Test that track_id is defined. This is not currently |
- // working since SSRCs are used to match RtpReceivers (and their tracks) with |
- // received stream stats in TrackMediaInfoMap. |
+ ASSERT_TRUE(inbound_stream_stats[0]->track_id.is_defined()); |
+} |
+ |
+// Test that we can successfully get the media related stats (audio level |
+// etc.) for the unsignaled stream. |
+TEST_F(PeerConnectionIntegrationTest, |
+ GetMediaStatsForUnsignaledStreamWithNewStatsApi) { |
+ ASSERT_TRUE(CreatePeerConnectionWrappers()); |
+ ConnectFakeSignaling(); |
+ caller()->AddAudioVideoMediaStream(); |
+ // Remove SSRCs and MSIDs from the received offer SDP. |
+ callee()->SetReceivedSdpMunger(RemoveSsrcsAndMsids); |
+ caller()->CreateAndSetAndSignalOffer(); |
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout); |
+ // Wait for one audio frame to be received by the callee. |
+ ExpectNewFramesReceivedWithWait(0, 0, 1, 1, kMaxWaitForFramesMs); |
+ |
+ rtc::scoped_refptr<const webrtc::RTCStatsReport> report = |
+ callee()->NewGetStats(); |
+ ASSERT_NE(nullptr, report); |
+ |
+ auto media_stats = report->GetStatsOfType<webrtc::RTCMediaStreamTrackStats>(); |
+ auto audio_index = FindFirstMediaStatsIndexByKind("audio", media_stats); |
+ ASSERT_GE(audio_index, 0); |
+ EXPECT_TRUE(media_stats[audio_index]->audio_level.is_defined()); |
} |
// Test that DTLS 1.0 is used if both sides only support DTLS 1.0. |