RtpPacket class introduced.

webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h

Index: webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
new file mode 100644
index 0000000000000000000000000000000000000000..ad749ffb61ec4f422ae9b04acd2698bc9862a8dd
--- /dev/null
+++ b/webrtc/modules/rtp_rtcp/source/rtp_packet_to_send.h
@@ -0,0 +1,33 @@
+/*
+ *  Copyright (c) 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_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+#define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_
+
+#include "webrtc/modules/rtp_rtcp/source/rtp_packet.h"
+
+namespace webrtc {
+// Class to hold rtp packet with metadata for sender side.
+class RtpPacketToSend : public rtp::Packet {
+ public:
+  explicit RtpPacketToSend(const ExtensionManager* extensions)
+      : Packet(extensions) {}
+  RtpPacketToSend(const ExtensionManager* extensions, size_t capacity)
+      : Packet(extensions, capacity) {}
+
+  // Time in local time base as close as it can to frame capture time.
+  int64_t capture_time_ms() const { return capture_time_ms_; }
+  void set_capture_time_ms(int64_t time) { capture_time_ms_ = time; }
+
+ private:
+  int64_t capture_time_ms_ = 0;
+};
+
+}  // namespace webrtc
+#endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_PACKET_TO_SEND_H_