Fix SDP stream ID mismatch issue when a track's stream changes.

webrtc/pc/peerconnection_integrationtest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 199 matching lines @@
         new FakeRTCCertificateGenerator());
     PeerConnectionWrapper* client(new PeerConnectionWrapper(debug_name));
     if (!client->Init(constraints, nullptr, nullptr, std::move(cert_generator),
                       network_thread, worker_thread)) {
       delete client;
       return nullptr;
     }
     return client;
   }
 
+  webrtc::PeerConnectionFactoryInterface* pc_factory() const {
+    return peer_connection_factory_.get();
+  }
+
   webrtc::PeerConnectionInterface* pc() const { return peer_connection_.get(); }
 
   // If a signaling message receiver is set (via ConnectFakeSignaling), this
   // will set the whole offer/answer exchange in motion. Just need to wait for
   // the signaling state to reach "stable".
   void CreateAndSetAndSignalOffer() {
     auto offer = CreateOffer();
     ASSERT_NE(nullptr, offer);
     EXPECT_TRUE(SetLocalDescriptionAndSendSdpMessage(std::move(offer)));
   }
@@ skipping 2536 matching lines @@
   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());
 }
 
+// Test that if a track is removed and added again with a different stream ID,
+// the new stream ID is successfully communicated in SDP and media continues to
+// flow end-to-end.
+TEST_F(PeerConnectionIntegrationTest, AddTrackWithDifferentStreamId) {
+  ASSERT_TRUE(CreatePeerConnectionWrappers());
+  ConnectFakeSignaling();
+
+  rtc::scoped_refptr<MediaStreamInterface> stream_1 =
+      caller()->pc_factory()->CreateLocalMediaStream("stream_1");
+  rtc::scoped_refptr<MediaStreamInterface> stream_2 =
+      caller()->pc_factory()->CreateLocalMediaStream("stream_2");
+
+  // Add track using stream 1, do offer/answer.
+  rtc::scoped_refptr<webrtc::AudioTrackInterface> track =
+      caller()->CreateLocalAudioTrack();
+  rtc::scoped_refptr<webrtc::RtpSenderInterface> sender =
+      caller()->pc()->AddTrack(track, {stream_1.get()});
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for one audio frame to be received by the callee.
+  ExpectNewFramesReceivedWithWait(0, 0, 1, 0, kMaxWaitForFramesMs);
+
+  // Remove the sender, and create a new one with the new stream.
+  caller()->pc()->RemoveTrack(sender);
+  sender = caller()->pc()->AddTrack(track, {stream_2.get()});
+  caller()->CreateAndSetAndSignalOffer();
+  ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
+  // Wait for additional audio frames to be received by the callee.
+  ExpectNewFramesReceivedWithWait(0, 0, kDefaultExpectedAudioFrameCount, 0,
+                                  kMaxWaitForFramesMs);
+}
+
 }  // namespace
 
 #endif  // if !defined(THREAD_SANITIZER)