Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

webrtc/call/rtp_transport_controller_receive.cc

Index: webrtc/call/rtp_transport_controller_receive.cc
diff --git a/webrtc/call/rtp_transport_controller_receive.cc b/webrtc/call/rtp_transport_controller_receive.cc
new file mode 100644
index 0000000000000000000000000000000000000000..377ea008c8a13fe475ff712c1b57baddaa2c571c
--- /dev/null
+++ b/webrtc/call/rtp_transport_controller_receive.cc
@@ -0,0 +1,56 @@
+/*
+ *  Copyright (c) 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/call/rtp_transport_controller_receive.h"
+#include "webrtc/base/ptr_util.h"
+
+namespace webrtc {
+
+RtpTransportControllerReceive::Receiver::Receiver(
+    RtpTransportControllerReceive* transport,
+    uint32_t ssrc,
+    RtpPacketSinkInterface* sink)
+    : transport_(transport), sink_(sink) {
+  transport_->AddSink(ssrc, sink_);
+}
+
+RtpTransportControllerReceive::Receiver::~Receiver() {
+  // Don't require return value > 0, since for RTX we currently may
+  // have multiple Receiver objects with the same sink.
+  // TODO(nisse): Consider adding a DCHECK when RtxReceiveStream is wired up.
+  transport_->RemoveSink(sink_);
+}
+
+std::unique_ptr<RtpTransportReceiver>
+RtpTransportControllerReceive::CreateReceiver(uint32_t ssrc,
+                                              RtpPacketSinkInterface* sink) {
+  return rtc::MakeUnique<Receiver>(this, ssrc, sink);
+}
+
+bool RtpTransportControllerReceive::OnRtpPacket(
+    const RtpPacketReceived& packet) {
+  rtc::CritScope cs(&lock_);
+  return demuxer_.OnRtpPacket(packet);
+}
+
+void RtpTransportControllerReceive::AddSink(uint32_t ssrc,
+                                            RtpPacketSinkInterface* sink) {
+  rtc::CritScope cs(&lock_);
+  return demuxer_.AddSink(ssrc, sink);
+}
+
+size_t RtpTransportControllerReceive::RemoveSink(
+    const RtpPacketSinkInterface* sink) {
+  rtc::CritScope cs(&lock_);
+  return demuxer_.RemoveSink(sink);
+}
+
+
+}  // namespace webrtc