Reland of Implemented the GetSources() in native code. (patchset #1 id:1 of https://codereview.webrtc.org/2809613002/ )
Reason for revert:
Re-land, reverting did not fix bug.
https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
Original issue's description:
> Revert of Implemented the GetSources() in native code. (patchset #11 id:510001 of https://codereview.webrtc.org/2770233003/ )
>
> Reason for revert:
> Suspected of WebRtcApprtcBrowserTest.MANUAL_WorksOnApprtc breakage, see
>
> https://bugs.chromium.org/p/webrtc/issues/detail?id=7465
>
> Original issue's description:
> > Added the GetSources() to the RtpReceiverInterface and implemented
> > it for the AudioRtpReceiver.
> >
> > This method returns a vector of RtpSource(both CSRC source and SSRC
> > source) which contains the ID of a source, the timestamp, the source
> > type (SSRC or CSRC) and the audio level.
> >
> > The RtpSource objects are buffered and maintained by the
> > RtpReceiver in webrtc/modules/rtp_rtcp/. When the method is called,
> > the info of the contributing source will be pulled along the object
> > chain:
> > AudioRtpReceiver -> VoiceChannel -> WebRtcVoiceMediaChannel ->
> > AudioReceiveStream -> voe::Channel -> RtpRtcp module
> >
> > Spec:https://w3c.github.io/webrtc-pc/archives/20151006/webrtc.html#widl-RTCRtpReceiver-getContributingSources-sequence-RTCRtpContributingSource
> >
> > BUG=chromium:703122
> > TBR=stefan@webrtc.org, danilchap@webrtc.org
> >
> > Review-Url: https://codereview.webrtc.org/2770233003
> > Cr-Commit-Position: refs/heads/master@{#17591}
> > Committed: https://chromium.googlesource.com/external/webrtc/+/292084c3765d9f3ee406ca2ec86eae206b540053
>
> TBR=deadbeef@webrtc.org,solenberg@webrtc.org,hbos@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org
> # Not skipping CQ checks because original CL landed more than 1 days ago.
> BUG=chromium:703122
>
> Review-Url: https://codereview.webrtc.org/2809613002
> Cr-Commit-Position: refs/heads/master@{#17616}
> Committed: https://chromium.googlesource.com/external/webrtc/+/fbcc5cb3869d1370008e40f24fc03ac8fb69c675
TBR=deadbeef@webrtc.org,solenberg@webrtc.org,philipel@webrtc.org,stefan@webrtc.org,danilchap@webrtc.org,zhihuang@webrtc.org,olka@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=chromium:703122
Review-Url: https://codereview.webrtc.org/2810623003
Cr-Commit-Position: refs/heads/master@{#17621}
diff --git a/webrtc/pc/peerconnection_integrationtest.cc b/webrtc/pc/peerconnection_integrationtest.cc
index b4acdaa..d818300 100644
--- a/webrtc/pc/peerconnection_integrationtest.cc
+++ b/webrtc/pc/peerconnection_integrationtest.cc
@@ -2755,6 +2755,24 @@
kMaxWaitForFramesMs);
}
+TEST_F(PeerConnectionIntegrationTest, GetSources) {
+ ASSERT_TRUE(CreatePeerConnectionWrappers());
+ ConnectFakeSignaling();
+ caller()->AddAudioOnlyMediaStream();
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+ // Wait for one audio frame received by callee.
+ ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
+ ASSERT_GT(callee()->pc()->GetReceivers().size(), 0u);
+ auto receiver = callee()->pc()->GetReceivers()[0];
+ ASSERT_EQ(receiver->media_type(), cricket::MEDIA_TYPE_AUDIO);
+
+ auto contributing_sources = receiver->GetSources();
+ ASSERT_GT(receiver->GetParameters().encodings.size(), 0u);
+ EXPECT_EQ(receiver->GetParameters().encodings[0].ssrc,
+ contributing_sources[0].source_id());
+}
+
} // namespace
#endif // if !defined(THREAD_SANITIZER)
