Introduce RtpTransportInternal and SrtpTransport.
SrtpTransport currently just delegates everything to RtpTransport.
Also makes BaseChannel::rtp_transport_ an RtpTransportInternal and constructs an SrtpTransport if srtp is required.
BUG=webrtc:7013
Review-Url: https://codereview.webrtc.org/2981013002
Cr-Commit-Position: refs/heads/master@{#19095}
diff --git a/webrtc/pc/srtptransport.cc b/webrtc/pc/srtptransport.cc
new file mode 100644
index 0000000..6e6ff06
--- /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
