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..aea414e08dfc2a162691a69171be0b7aab909ce3
--- /dev/null
+++ b/webrtc/call/rtp_demuxer.cc
@@ -0,0 +1,52 @@
+/*
+ *  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/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) {
+  RTC_DCHECK(sink);
+  sinks_.emplace(ssrc, sink);
+}
+
+size_t RtpDemuxer::RemoveSink(const RtpPacketSinkInterface* sink) {
+  RTC_DCHECK(sink);
+  size_t count = 0;
+  for (auto it = sinks_.begin(); it != sinks_.end(); ) {
+    if (it->second == sink) {
+      it = sinks_.erase(it);
+      ++count;
+    } else {
+      ++it;
+    }
+  }
+  return count;
+}
+
+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