Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/rtptransport_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.
  */
 
 #include <string>
 
 #include "webrtc/p2p/base/fakepackettransport.h"
 #include "webrtc/pc/rtptransport.h"
+#include "webrtc/pc/rtptransporttestutil.h"
 #include "webrtc/rtc_base/gunit.h"
 
 namespace webrtc {
 
 constexpr bool kMuxDisabled = false;
 constexpr bool kMuxEnabled = true;
 
 TEST(RtpTransportTest, SetRtcpParametersCantDisableRtcpMux) {
   RtpTransport transport(kMuxDisabled);
   RtcpParameters params;
@@ skipping 119 matching lines @@
 
   // State does not change, so we should not signal.
   transport.SetRtcpMuxEnabled(true);
   EXPECT_EQ(observer.count(), 1);
 
   // State changes, so we should signal.
   transport.SetRtcpMuxEnabled(false);
   EXPECT_EQ(observer.count(), 2);
 }
 
-class SignalPacketReceivedCounter : public sigslot::has_slots<> {
- public:
-  explicit SignalPacketReceivedCounter(RtpTransport* transport) {
-    transport->SignalPacketReceived.connect(
-        this, &SignalPacketReceivedCounter::OnPacketReceived);
-  }
-  int rtcp_count() const { return rtcp_count_; }
-  int rtp_count() const { return rtp_count_; }
-
- private:
-  void OnPacketReceived(bool rtcp,
-                        rtc::CopyOnWriteBuffer*,
-                        const rtc::PacketTime&) {
-    if (rtcp) {
-      ++rtcp_count_;
-    } else {
-      ++rtp_count_;
-    }
-  }
-  int rtcp_count_ = 0;
-  int rtp_count_ = 0;
-};
-
 // Test that SignalPacketReceived fires with rtcp=true when a RTCP packet is
 // received.
 TEST(RtpTransportTest, SignalDemuxedRtcp) {
   RtpTransport transport(kMuxDisabled);
   SignalPacketReceivedCounter observer(&transport);
   rtc::FakePacketTransport fake_rtp("fake_rtp");
   fake_rtp.SetDestination(&fake_rtp, true);
   transport.SetRtpPacketTransport(&fake_rtp);
 
   // An rtcp packet.
@@ skipping 40 matching lines @@
 
   const rtc::PacketOptions options;
   const int flags = 0;
   rtc::Buffer rtp_data(kRtpData, kRtpLen);
   fake_rtp.SendPacket(rtp_data.data<char>(), kRtpLen, options, flags);
   EXPECT_EQ(0, observer.rtp_count());
   EXPECT_EQ(0, observer.rtcp_count());
 }
 
 }  // namespace webrtc