Adding regression test for rejecting and un-rejecting an m= section.
This was previously not working because the answerer wasn't generating
ICE credentials when it should have been.
This was fixed inadvertently by:
https://webrtc-review.googlesource.com/c/src/+/46380
But we should really also have a PeerConnection-level regression test
for this.
Bug: webrtc:6023
Change-Id: I3da900edcc8db8034ed61a7bb981d9c0e616254e
Reviewed-on: https://webrtc-review.googlesource.com/69403
Commit-Queue: Taylor Brandstetter <deadbeef@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#22832}diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index b9a0227..1249298 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -2176,6 +2176,43 @@
}
}
+// Do one offer/answer with audio, another that disables it (rejecting the m=
+// section), and another that re-enables it. Regression test for:
+// bugs.webrtc.org/6023
+TEST_F(PeerConnectionIntegrationTestPlanB, EnableAudioAfterRejecting) {
+ ASSERT_TRUE(CreatePeerConnectionWrappers());
+ ConnectFakeSignaling();
+
+ // Add audio track, do normal offer/answer.
+ rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+ caller()->CreateLocalAudioTrack();
+ rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+ caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+ // Remove audio track, and set offer_to_receive_audio to false to cause the
+ // m= section to be completely disabled, not just "recvonly".
+ caller()->pc()->RemoveTrack(sender);
+ PeerConnectionInterface::RTCOfferAnswerOptions options;
+ options.offer_to_receive_audio = 0;
+ caller()->SetOfferAnswerOptions(options);
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+ // Add the audio track again, expecting negotiation to succeed and frames to
+ // flow.
+ sender = caller()->pc()->AddTrack(track, {"stream"}).MoveValue();
+ options.offer_to_receive_audio = 1;
+ caller()->SetOfferAnswerOptions(options);
+ caller()->CreateAndSetAndSignalOffer();
+ ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+
+ MediaExpectations media_expectations;
+ media_expectations.CalleeExpectsSomeAudio();
+ EXPECT_TRUE(ExpectNewFrames(media_expectations));
+}
+
// Basic end-to-end test, but without SSRC/MSID signaling. This functionality
// is needed to support legacy endpoints.
// TODO(deadbeef): When we support the MID extension and demuxing on MID, also
