[Draft] RtpPacket sketched.

webrtc/modules/rtp_rtcp/include/rtp_rtcp.h

 /*
  *  Copyright (c) 2012 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_INCLUDE_RTP_RTCP_H_
 #define WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_
 
 #include <set>
 #include <string>
 #include <utility>
 #include <vector>
 
+#include "webrtc/call.h"
 #include "webrtc/modules/include/module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/video_coding/include/video_coding_defines.h"
 
 namespace webrtc {
 // Forward declarations.
 class ReceiveStatistics;
 class RemoteBitrateEstimator;
 class RtpReceiver;
 class Transport;
@@ skipping 152 matching lines @@
    /*
     *   (De)register RTP header extension type and id.
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RegisterSendRtpHeaderExtension(RTPExtensionType type,
                                                    uint8_t id) = 0;
 
     virtual int32_t DeregisterSendRtpHeaderExtension(RTPExtensionType type) = 0;
 
+    // TODO(danilchap): Make pure virtual in 3 months.
+    virtual bool RegisterRtpHeaderExtension(const std::string& type,
+                                            MediaType media,
+                                            uint8_t id);
+
     /*
     *   get start timestamp
     */
     virtual uint32_t StartTimestamp() const = 0;
 
     /*
     *   configure start timestamp, default is a random number
     *
     *   timestamp   - start timestamp
     */
@@ skipping 154 matching lines @@
     *   return -1 on failure else 0
     */
     virtual int32_t RemoteCNAME(uint32_t remoteSSRC,
                                 char cName[RTCP_CNAME_SIZE]) const = 0;
 
     /*
     *   Get remote NTP
     *
     *   return -1 on failure else 0
     */
-    virtual int32_t RemoteNTP(
-        uint32_t *ReceivedNTPsecs,
-        uint32_t *ReceivedNTPfrac,
-        uint32_t *RTCPArrivalTimeSecs,
-        uint32_t *RTCPArrivalTimeFrac,
-        uint32_t *rtcp_timestamp) const  = 0;
+    virtual int32_t RemoteNTP(uint32_t* ReceivedNTPsecs,
+                              uint32_t* ReceivedNTPfrac,
+                              uint32_t* RTCPArrivalTimeSecs,
+                              uint32_t* RTCPArrivalTimeFrac,
+                              uint32_t* rtcp_timestamp) const = 0;
 
     /*
     *   AddMixedCNAME
     *
     *   return -1 on failure else 0
     */
     virtual int32_t AddMixedCNAME(uint32_t SSRC, const char* c_name) = 0;
 
     /*
     *   RemoveMixedCNAME
@@ skipping 160 matching lines @@
     *   Send NACK for the packets specified.
     */
     virtual void SendNack(const std::vector<uint16_t>& sequence_numbers) = 0;
 
     /*
     *   Store the sent packets, needed to answer to a Negative acknowledgement
     *   requests
     */
     virtual void SetStorePacketsStatus(bool enable, uint16_t numberToStore) = 0;
 
-    // Returns true if the module is configured to store packets.
-    virtual bool StorePackets() const = 0;
-
     // Called on receipt of RTCP report block from remote side.
     virtual void RegisterRtcpStatisticsCallback(
         RtcpStatisticsCallback* callback) = 0;
-    virtual RtcpStatisticsCallback*
-        GetRtcpStatisticsCallback() = 0;
+    virtual RtcpStatisticsCallback* GetRtcpStatisticsCallback() = 0;
     // BWE feedback packets.
     virtual bool SendFeedbackPacket(const rtcp::TransportFeedback& packet) = 0;
 
     /**************************************************************************
     *
     *   Audio
     *
     ***************************************************************************/
 
     /*
@@ skipping 88 matching lines @@
 
     /*
     *   send a request for a keyframe
     *
     *   return -1 on failure else 0
     */
     virtual int32_t RequestKeyFrame() = 0;
 };
 }  // namespace webrtc
 #endif  // WEBRTC_MODULES_RTP_RTCP_INCLUDE_RTP_RTCP_H_