Follow standard sending CVO rtp header extension

webrtc/modules/rtp_rtcp/source/rtp_sender_video.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_SOURCE_RTP_SENDER_VIDEO_H_
 #define WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_
 
 #include <list>
 #include <memory>
 
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/onetimeevent.h"
 #include "webrtc/base/rate_statistics.h"
+#include "webrtc/base/sequenced_task_checker.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/forward_error_correction.h"
 #include "webrtc/modules/rtp_rtcp/source/producer_fec.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_rtcp_config.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_sender.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/video_codec_information.h"
 #include "webrtc/typedefs.h"
@@ skipping 52 matching lines @@
 
   void SendVideoPacketAsRed(std::unique_ptr<RtpPacketToSend> media_packet,
                             StorageType media_packet_storage,
                             bool protect);
 
   RTPSender* const rtp_sender_;
   Clock* const clock_;
 
   // Should never be held when calling out of this class.
   rtc::CriticalSection crit_;
+  rtc::SequencedTaskChecker encoder_checker_;
 
   RtpVideoCodecTypes video_type_ = kRtpVideoGeneric;
   int32_t retransmission_settings_ GUARDED_BY(crit_) = kRetransmitBaseLayer;
+  VideoRotation last_rotation_ GUARDED_BY(encoder_checker_) = kVideoRotation_0;
 
   // FEC
   bool fec_enabled_ GUARDED_BY(crit_) = false;
   int8_t red_payload_type_ GUARDED_BY(crit_) = 0;
   int8_t fec_payload_type_ GUARDED_BY(crit_) = 0;
   FecProtectionParams delta_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{
       0, 1, kFecMaskRandom};
   FecProtectionParams key_fec_params_ GUARDED_BY(crit_) = FecProtectionParams{
       0, 1, kFecMaskRandom};
   ProducerFec producer_fec_ GUARDED_BY(crit_);
 
   rtc::CriticalSection stats_crit_;
   // Bitrate used for FEC payload, RED headers, RTP headers for FEC packets
   // and any padding overhead.
   RateStatistics fec_bitrate_ GUARDED_BY(stats_crit_);
   // Bitrate used for video payload and RTP headers.
   RateStatistics video_bitrate_ GUARDED_BY(stats_crit_);
   OneTimeEvent first_frame_sent_;
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_RTP_RTCP_SOURCE_RTP_SENDER_VIDEO_H_