[rtp_rtcp] rtcp::Sli packet moved into own file and got Parse function

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.
  */
 
 
 /*
  * This file includes unit tests for the RTCPReceiver.
  */
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/common_types.h"
 #include "webrtc/modules/remote_bitrate_estimator/include/mock/mock_remote_bitrate_observer.h"
 #include "webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/app.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/pli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sli.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 
 namespace webrtc {
 
 namespace {  // Anonymous namespace; hide utility functions and classes.
 
@@ skipping 521 matching lines @@
 
   rtcp::Fir fir;
   fir.To(kSourceSsrc + 1);
   rtc::scoped_ptr<rtcp::RawPacket> packet(fir.Build());
   EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(0U, rtcp_packet_info_.rtcpPacketTypeFlags);
 }
 
 TEST_F(RtcpReceiverTest, InjectSliPacket) {
   rtcp::Sli sli;
-  sli.WithPictureId(40);
+  sli.WithMacroblocks(40, 0, 1);
   rtc::scoped_ptr<rtcp::RawPacket> packet(sli.Build());
   EXPECT_EQ(0, InjectRtcpPacket(packet->Buffer(), packet->Length()));
   EXPECT_EQ(kRtcpSli, rtcp_packet_info_.rtcpPacketTypeFlags);
   EXPECT_EQ(40, rtcp_packet_info_.sliPictureId);
 }
 
 TEST_F(RtcpReceiverTest, XrPacketWithZeroReportBlocksIgnored) {
   rtcp::Xr xr;
   xr.From(0x2345);
   rtc::scoped_ptr<rtcp::RawPacket> packet(xr.Build());
@@ skipping 512 matching lines @@
 
   // Transport feedback should be ignored, but next packet should work.
   EXPECT_EQ(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpTransportFeedback);
   EXPECT_NE(0u, rtcp_packet_info_.rtcpPacketTypeFlags & kRtcpRemb);
   EXPECT_EQ(kBitrateBps, rtcp_packet_info_.receiverEstimatedMaxBitrate);
 }
 
 }  // Anonymous namespace
 
 }  // namespace webrtc