Add rtppacketuil.h/cc

webrtc/p2p/base/rtppacketutil.h

+/*
+ *  Copyright 2016 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.
+ */
+
+#ifndef WEBRTC_P2P_BASE_RTPPACKETUTIL_H_
+#define WEBRTC_P2P_BASE_RTPPACKETUTIL_H_
+
+#include <cstddef>
+#include <cstdint>
+
+namespace rtc {
+
+struct PacketOptions;
+
+// Returns true if the packet in |data| is a DTLS packet.
+bool IsDtlsPacket(const char* data, size_t length);
+
+// Returns true if the packet in |data| is an RTCP packet.
+bool IsRtcpPacket(const char* data, size_t length);
+
+// Applies specified |options| to the packet. It updates absolute send time
+// extension header if it is present present then updates HMAC.
+// If |abs_send_time| is 0 then current system time is used.
+bool ApplyPacketOptions(char* data,
+                        size_t length,
+                        const PacketOptions& options,
+                        uint32_t abs_send_time);
+
+// Verifies that a packet has a valid RTP header.
+bool ValidateRtpHeader(const char* rtp, size_t length, size_t* header_length);
+
+// Finds RTP offset and length if the packet is encapsulated in a TURN Channel
+// Message or TURN Send Indication message.
+bool GetRtpPacketStartPositionAndLength(const char* data,
+                                        size_t length,
+                                        size_t* rtp_start_pos,
+                                        size_t* rtp_packet_length);
+
+// Helper method which updates absolute send time extension if present.
+bool UpdateRtpAbsSendTimeExtension(char* rtp,
+                                   size_t length,
+                                   int extension_id,
+                                   uint32_t abs_send_time);
+
+}  // namespace rtc
+
+#endif  // WEBRTC_P2P_BASE_RTPPACKETUTIL_H_