New class RtpDemuxer and RtpPacketSinkInterface, use in Call.

webrtc/call/rtp_demuxer.cc

Index: webrtc/call/rtp_demuxer.cc
diff --git a/webrtc/call/rtp_demuxer.cc b/webrtc/call/rtp_demuxer.cc
new file mode 100644
index 0000000000000000000000000000000000000000..638c8c6e2ab5d08e23f5cf4aae65086c3c4497bd
--- /dev/null
+++ b/webrtc/call/rtp_demuxer.cc
@@ -0,0 +1,48 @@
+/*
+ *  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 <utility>
+
+#include "webrtc/base/checks.h"
+#include "webrtc/call/rtp_demuxer.h"
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
+
+namespace webrtc {
+
+RtpDemuxer::RtpDemuxer() {}
+
+RtpDemuxer::~RtpDemuxer() {
+  RTC_DCHECK(sinks_.empty());
+}
+
+void RtpDemuxer::AddSink(uint32_t ssrc, RtpPacketSinkInterface* sink) {
+  sinks_.insert(std::pair<uint32_t, RtpPacketSinkInterface*>(ssrc, sink));

[danilchap, 2017/05/09 09:20:22]

sinks_.emplace(ssrc, sink);

[nisse-webrtc, 2017/05/09 12:11:49]

Thanks, that looks better.

+}
+
+void RtpDemuxer::RemoveSink(const RtpPacketSinkInterface* sink) {
+  for (auto it = sinks_.begin(); it != sinks_.end(); ) {
+    if (it->second == sink)
+      it = sinks_.erase(it);
+    else
+      ++it;
+  }
+}
+
+bool RtpDemuxer::OnRtpPacket(const RtpPacketReceived& packet) {
+  bool found = false;
+  auto it_range = sinks_.equal_range(packet.Ssrc());
+  for (auto it = it_range.first; it != it_range.second; ++it) {
+    found = true;
+    it->second->OnRtpPacket(packet);
+  }
+  return found;
+}
+
+}  // namespace webrtc