Follow standard sending CVO rtp header extension

webrtc/modules/rtp_rtcp/source/rtp_sender_video.cc

 /*
  *  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.
  */
 
@@ skipping 31 matching lines @@
   red_payload[0] = media_packet.PayloadType();
   memcpy(&red_payload[kRedForFecHeaderLength], media_packet.payload(),
          media_packet.payload_size());
 }
 }  // namespace
 
 RTPSenderVideo::RTPSenderVideo(Clock* clock, RTPSender* rtp_sender)
     : rtp_sender_(rtp_sender),
       clock_(clock),
       fec_bitrate_(1000, RateStatistics::kBpsScale),
-      video_bitrate_(1000, RateStatistics::kBpsScale) {}
+      video_bitrate_(1000, RateStatistics::kBpsScale) {
+  encoder_checker_.Detach();
+}
 
 RTPSenderVideo::~RTPSenderVideo() {}
 
 void RTPSenderVideo::SetVideoCodecType(RtpVideoCodecTypes video_type) {
   video_type_ = video_type;
 }
 
 RtpVideoCodecTypes RTPSenderVideo::VideoCodecType() const {
   return video_type_;
 }
@@ skipping 158 matching lines @@
 
 bool RTPSenderVideo::SendVideo(RtpVideoCodecTypes video_type,
                                FrameType frame_type,
                                int8_t payload_type,
                                uint32_t rtp_timestamp,
                                int64_t capture_time_ms,
                                const uint8_t* payload_data,
                                size_t payload_size,
                                const RTPFragmentationHeader* fragmentation,
                                const RTPVideoHeader* video_header) {
+  RTC_DCHECK_CALLED_SEQUENTIALLY(&encoder_checker_);
   if (payload_size == 0)
     return false;
 
   // Create header that will be reused in all packets.
   std::unique_ptr<RtpPacketToSend> rtp_header = rtp_sender_->AllocatePacket();
   rtp_header->SetPayloadType(payload_type);
   rtp_header->SetTimestamp(rtp_timestamp);
   rtp_header->set_capture_time_ms(capture_time_ms);
   // According to
   // http://www.etsi.org/deliver/etsi_ts/126100_126199/126114/12.07.00_60/
   // ts_126114v120700p.pdf Section 7.4.5:
   // The MTSI client shall add the payload bytes as defined in this clause
   // onto the last RTP packet in each group of packets which make up a key
   // frame (I-frame or IDR frame in H.264 (AVC), or an IRAP picture in H.265
   // (HEVC)). The MTSI client may also add the payload bytes onto the last RTP
   // packet in each group of packets which make up another type of frame
   // (e.g. a P-Frame) only if the current value is different from the previous
   // value sent.
-  // Here we are adding it to every packet of every frame at this point.
-  if (video_header && video_header->rotation != kVideoRotation_0)
-    rtp_header->SetExtension<VideoOrientation>(video_header->rotation);
+  if (video_header) {
+    // Set rotation when key frame or when changed (to follow standard).
+    // Or when different from 0 (to follow current receiver implementation).
+    VideoRotation current_rotation = video_header->rotation;
+    if (frame_type == kVideoFrameKey || current_rotation != last_rotation_ ||
+        current_rotation != kVideoRotation_0)
+      rtp_header->SetExtension<VideoOrientation>(current_rotation);
+    last_rotation_ = current_rotation;
+  }
 
   size_t packet_capacity = rtp_sender_->MaxPayloadLength() -
                            FecPacketOverhead() -
                            (rtp_sender_->RtxStatus() ? kRtxHeaderSize : 0);
   RTC_DCHECK_LE(packet_capacity, rtp_header->capacity());
   RTC_DCHECK_GT(packet_capacity, rtp_header->headers_size());
   size_t max_data_payload_length = packet_capacity - rtp_header->headers_size();
 
   std::unique_ptr<RtpPacketizer> packetizer(RtpPacketizer::Create(
       video_type, max_data_payload_length,
@@ skipping 74 matching lines @@
   rtc::CritScope cs(&crit_);
   return retransmission_settings_;
 }
 
 void RTPSenderVideo::SetSelectiveRetransmissions(uint8_t settings) {
   rtc::CritScope cs(&crit_);
   retransmission_settings_ = settings;
 }
 
 }  // namespace webrtc