Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/video/video_receive_stream_unittest.cc

Index: webrtc/video/video_receive_stream_unittest.cc
diff --git a/webrtc/video/video_receive_stream_unittest.cc b/webrtc/video/video_receive_stream_unittest.cc
index 237eed40b86074082b6ddb5804d8ac8a4d9404b1..c7f54408657bdac64cf36fda1efbe55d26d0b26e 100644
--- a/webrtc/video/video_receive_stream_unittest.cc
+++ b/webrtc/video/video_receive_stream_unittest.cc
@@ -16,6 +16,7 @@
 #include "webrtc/api/video_codecs/video_decoder.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
+#include "webrtc/call/rtp_stream_receiver_controller.h"
 #include "webrtc/media/base/fakevideorenderer.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h"
@@ -28,12 +29,11 @@
 using testing::_;

[danilchap, 2017/06/13 15:58:32]

move these 2 usings inside unnamed namespace too (since you are expanding it
anyway)

[nisse-webrtc, 2017/06/14 06:31:06]

Done.

 using testing::Invoke;
 
-constexpr int kDefaultTimeOutMs = 50;
-
 namespace webrtc {
-
 namespace {
 
+constexpr int kDefaultTimeOutMs = 50;
+
 const char kNewJitterBufferFieldTrialEnabled[] =
     "WebRTC-NewVideoJitterBuffer/Enabled/";
 
@@ -91,7 +91,7 @@ class VideoReceiveStreamTest : public testing::Test {
     config_.decoders.push_back(null_decoder);
 
     video_receive_stream_.reset(new webrtc::internal::VideoReceiveStream(
-        kDefaultNumCpuCores,
+        &rtp_stream_receiver_controller_, kDefaultNumCpuCores,
         &packet_router_, config_.Copy(), process_thread_.get(), &call_stats_));
   }
 
@@ -105,6 +105,7 @@ class VideoReceiveStreamTest : public testing::Test {
   MockTransport mock_transport_;
   PacketRouter packet_router_;
   std::unique_ptr<ProcessThread> process_thread_;
+  RtpStreamReceiverController rtp_stream_receiver_controller_;
   std::unique_ptr<webrtc::internal::VideoReceiveStream> video_receive_stream_;
 };
 
@@ -130,9 +131,10 @@ TEST_F(VideoReceiveStreamTest, CreateFrameFromH264FmtpSpropAndIdr) {
   EXPECT_CALL(mock_h264_video_decoder_, Decode(_, false, _, _, _));
   RtpPacketReceived parsed_packet;
   ASSERT_TRUE(parsed_packet.Parse(rtppacket.data(), rtppacket.size()));
-  video_receive_stream_->OnRtpPacket(parsed_packet);
+  rtp_stream_receiver_controller_.OnRtpPacket(parsed_packet);
   EXPECT_CALL(mock_h264_video_decoder_, Release());
   // Make sure the decoder thread had a chance to run.
   init_decode_event_.Wait(kDefaultTimeOutMs);
 }
+
 }  // namespace webrtc