commit 49ac5959c28eb4a6ef12ab6213e624f9da11949b
author Jonas Oreland <jonaso@webrtc.org> Wed Sep 26 14:04:32 2018
committer Commit Bot <commit-bot@chromium.org> Thu Sep 27 10:00:40 2018
tree a9f018b1798cdedad38f77aefaed68907cf07772
parent 3c7694137a0e868e172fa38dac092f09c591f0a1

Add GetSources to VideoRtpReceiver

BUG=webrtc:9770

Change-Id: I16143fce6eb727bbab0f6c621aa5b51bc6d28d6b
Reviewed-on: https://webrtc-review.googlesource.com/101600
Reviewed-by: Seth Hampson <shampson@webrtc.org>
Reviewed-by: Åsa Persson <asapersson@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Reviewed-by: Niels Moller <nisse@webrtc.org>
Commit-Queue: Jonas Oreland <jonaso@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24858}

pc/peerconnection_integrationtest.cc

diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index dd977b4..3cf3e58 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -4308,7 +4308,7 @@
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
 }
 
-TEST_P(PeerConnectionIntegrationTest, GetSources) {
+TEST_P(PeerConnectionIntegrationTest, GetSourcesAudio) {
   ASSERT_TRUE(CreatePeerConnectionWrappers());
   ConnectFakeSignaling();
   caller()->AddAudioTrack();
@@ -4318,14 +4318,34 @@
   MediaExpectations media_expectations;
   media_expectations.CalleeExpectsSomeAudio(1);
   ASSERT_TRUE(ExpectNewFrames(media_expectations));
-  ASSERT_GT(callee()->pc()->GetReceivers().size(), 0u);
+  ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
   auto receiver = callee()->pc()->GetReceivers()[0];
   ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
-
-  auto contributing_sources = receiver->GetSources();
+  auto sources = receiver->GetSources();
   ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
   EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
-            contributing_sources[0].source_id());
+            sources[0].source_id());
+  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
+}
+
+TEST_P(PeerConnectionIntegrationTest, GetSourcesVideo) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+  caller()->AddVideoTrack();
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for one video frame to be received by the callee.
+  MediaExpectations media_expectations;
+  media_expectations.CalleeExpectsSomeVideo(1);
+  ASSERT_TRUE(ExpectNewFrames(media_expectations));
+  ASSERT_EQ(callee()->pc()->GetReceivers().size(), 1u);
+  auto receiver = callee()->pc()->GetReceivers()[0];
+  ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_VIDEO);
+  auto sources = receiver->GetSources();
+  ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
+  EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
+            sources[0].source_id());
+  EXPECT_EQ(webrtc::RtpSourceType::SSRC, sources[0].source_type());
 }
 
 // Test that if a track is removed and added again with a different stream ID,