Introduce RtpTransportInternal and SrtpTransport.

webrtc/pc/srtptransport.cc

Index: webrtc/pc/srtptransport.cc
diff --git a/webrtc/pc/srtptransport.cc b/webrtc/pc/srtptransport.cc
new file mode 100644
index 0000000000000000000000000000000000000000..6e6ff062749ba8f6d8cac18ef9484b27ac0df4c1
--- /dev/null
+++ b/webrtc/pc/srtptransport.cc
@@ -0,0 +1,62 @@
+/*
+ *  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 "webrtc/pc/srtptransport.h"
+
+#include <string>
+
+#include "webrtc/media/base/rtputils.h"
+#include "webrtc/pc/rtptransport.h"
+#include "webrtc/pc/srtpsession.h"
+#include "webrtc/rtc_base/asyncpacketsocket.h"
+#include "webrtc/rtc_base/copyonwritebuffer.h"
+#include "webrtc/rtc_base/ptr_util.h"
+#include "webrtc/rtc_base/trace_event.h"
+
+namespace webrtc {
+
+SrtpTransport::SrtpTransport(bool rtcp_mux_enabled,
+                             const std::string& content_name)
+    : content_name_(content_name),
+      rtp_transport_(rtc::MakeUnique<RtpTransport>(rtcp_mux_enabled)) {
+  ConnectToRtpTransport();
+}
+
+SrtpTransport::SrtpTransport(std::unique_ptr<RtpTransportInternal> transport,
+                             const std::string& content_name)
+    : content_name_(content_name), rtp_transport_(std::move(transport)) {
+  ConnectToRtpTransport();
+}
+
+void SrtpTransport::ConnectToRtpTransport() {
+  rtp_transport_->SignalPacketReceived.connect(
+      this, &SrtpTransport::OnPacketReceived);
+  rtp_transport_->SignalReadyToSend.connect(this,
+                                            &SrtpTransport::OnReadyToSend);
+}
+
+bool SrtpTransport::SendPacket(bool rtcp,
+                               rtc::CopyOnWriteBuffer* packet,
+                               const rtc::PacketOptions& options,
+                               int flags) {
+  // TODO(zstein): Protect packet.
+
+  return rtp_transport_->SendPacket(rtcp, packet, options, flags);
+}
+
+void SrtpTransport::OnPacketReceived(bool rtcp,
+                                     rtc::CopyOnWriteBuffer* packet,
+                                     const rtc::PacketTime& packet_time) {
+  // TODO(zstein): Unprotect packet.
+
+  SignalPacketReceived(rtcp, packet, packet_time);
+}
+
+}  // namespace webrtc