Create RtcpDemuxer

webrtc/call/rtp_rtcp_demuxer_helper.cc

Index: webrtc/call/rtp_rtcp_demuxer_helper.cc
diff --git a/webrtc/call/rtp_rtcp_demuxer_helper.cc b/webrtc/call/rtp_rtcp_demuxer_helper.cc
new file mode 100644
index 0000000000000000000000000000000000000000..1749e5497c14f439bd94f934c2593fc739e07f98
--- /dev/null
+++ b/webrtc/call/rtp_rtcp_demuxer_helper.cc
@@ -0,0 +1,50 @@
+/*
+ *  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_rtcp_demuxer_helper.h"
+
+#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/bye.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/psfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rtpfb.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/sender_report.h"
+
+namespace webrtc {
+
+rtc::Optional<uint32_t> ParseRtcpPacketSenderSsrc(
+    rtc::ArrayView<const uint8_t> packet) {
+  rtcp::CommonHeader header;
+  for (const uint8_t* next_packet = packet.begin(); next_packet != packet.end();
+       next_packet = header.NextPacket()) {
+    if (!header.Parse(next_packet, packet.end() - next_packet)) {
+      return rtc::Optional<uint32_t>();
+    }
+
+    if (header.payload_size_bytes() >= sizeof(uint32_t)) {
+      switch (header.type()) {
+        case rtcp::Bye::kPacketType:
+        case rtcp::Psfb::kPacketType:
+        case rtcp::ReceiverReport::kPacketType:
+        case rtcp::Rtpfb::kPacketType:
+        case rtcp::SenderReport::kPacketType:
+          // Sender SSRC at the beginning of the RTCP payload.
+          const uint32_t ssrc_sender =
+              webrtc::ByteReader<uint32_t>::ReadBigEndian(header.payload());

[danilchap, 2017/06/19 13:31:56]

drop webrtc:: prefix

[eladalon, 2017/06/19 14:35:17]

Done.

+          return rtc::Optional<uint32_t>(ssrc_sender);
+      }
+    }
+  }
+
+  return rtc::Optional<uint32_t>();
+}
+
+}  // namespace webrtc