Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/video/video_receive_stream_unittest.cc

 /*
  *  Copyright 2017 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 11 matching lines @@
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_receive_stream.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/sleep.h"
 #include "webrtc/test/field_trial.h"
 
 using testing::_;
 using testing::Invoke;
 
+namespace webrtc {
+#if 0
+namespace {
+
 constexpr int kDefaultTimeOutMs = 50;
 
-namespace webrtc {
-
-namespace {
-
 const char kNewJitterBufferFieldTrialEnabled[] =
     "WebRTC-NewVideoJitterBuffer/Enabled/";
 
 class MockTransport : public Transport {
  public:
   MOCK_METHOD3(SendRtp,
                bool(const uint8_t* packet,
                     size_t length,
                     const PacketOptions& options));
   MOCK_METHOD2(SendRtcp, bool(const uint8_t* packet, size_t length));
@@ skipping 81 matching lines @@
   EXPECT_CALL(mock_h264_video_decoder_, RegisterDecodeCompleteCallback(_));
   video_receive_stream_->Start();
   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);
   EXPECT_CALL(mock_h264_video_decoder_, Release());
   // Make sure the decoder thread had a chance to run.
   init_decode_event_.Wait(kDefaultTimeOutMs);
 }
+#endif
+
 }  // namespace webrtc