Verify sender ssrc when receiving rtcp target bitrate

webrtc/modules/rtp_rtcp/source/rtcp_receiver_unittest.cc

 /*
  *  Copyright (c) 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 1207 matching lines @@
 
   rtcp::TargetBitrate bitrate;
   bitrate.AddTargetBitrate(0, 0, expected_allocation.GetBitrate(0, 0) / 1000);
   bitrate.AddTargetBitrate(0, 1, expected_allocation.GetBitrate(0, 1) / 1000);
   bitrate.AddTargetBitrate(1, 0, expected_allocation.GetBitrate(1, 0) / 1000);
   bitrate.AddTargetBitrate(1, 1, expected_allocation.GetBitrate(1, 1) / 1000);
 
   rtcp::ExtendedReports xr;
   xr.SetTargetBitrate(bitrate);
 
+  // Wrong sender ssrc, target bitrate should be discarded.
+  xr.SetSenderSsrc(kSenderSsrc + 1);
+  EXPECT_CALL(bitrate_allocation_observer_,
+              OnBitrateAllocationUpdated(expected_allocation))
+      .Times(0);
+  InjectRtcpPacket(xr);
+
+  // Set correct ssrc, callback should be called once.
+  xr.SetSenderSsrc(kSenderSsrc);
   EXPECT_CALL(bitrate_allocation_observer_,
               OnBitrateAllocationUpdated(expected_allocation));
   InjectRtcpPacket(xr);
 }
 
 TEST_F(RtcpReceiverTest, HandlesIncorrectTargetBitrate) {
   BitrateAllocation expected_allocation;
   expected_allocation.SetBitrate(0, 0, 10000);
 
   rtcp::TargetBitrate bitrate;
   bitrate.AddTargetBitrate(0, 0, expected_allocation.GetBitrate(0, 0) / 1000);
   bitrate.AddTargetBitrate(0, kMaxTemporalStreams, 20000);
   bitrate.AddTargetBitrate(kMaxSpatialLayers, 0, 40000);
 
   rtcp::ExtendedReports xr;
   xr.SetTargetBitrate(bitrate);
+  xr.SetSenderSsrc(kSenderSsrc);
 
   EXPECT_CALL(bitrate_allocation_observer_,
               OnBitrateAllocationUpdated(expected_allocation));
   InjectRtcpPacket(xr);
 }
 
 }  // namespace webrtc