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Adding regression test for rejecting and un-rejecting an m= section.

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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}
This commit is contained in:
Taylor Brandstetter 2018-04-11 15:20:27 -07:00 committed by Commit Bot
parent 7435b84b30
commit 60c8dc8470
1 changed files with 37 additions and 0 deletions
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37
pc/peerconnection_integrationtest.cc
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@ -2176,6 +2176,43 @@ TEST_P(PeerConnectionIntegrationTest, VideoRejectedInSubsequentOffer) {
}
}
// 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