Add content type information to encoded images and corresponding rtp extension header

webrtc/video/video_send_stream.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include "webrtc/video/video_send_stream.h"
@@ skipping 616 matching lines @@
     const std::map<uint32_t, RtpState>& suspended_ssrcs)
     : worker_queue_(worker_queue),
       thread_sync_event_(false /* manual_reset */, false),
       stats_proxy_(Clock::GetRealTimeClock(),
                    config,
                    encoder_config.content_type),
       config_(std::move(config)),
       content_type_(encoder_config.content_type) {
   vie_encoder_.reset(new ViEEncoder(
       num_cpu_cores, &stats_proxy_, config_.encoder_settings,
-      config_.pre_encode_callback, config_.post_encode_callback));
+      config_.pre_encode_callback, config_.post_encode_callback,
+      content_type_));
   worker_queue_->PostTask(std::unique_ptr<rtc::QueuedTask>(new ConstructionTask(
       &send_stream_, &thread_sync_event_, &stats_proxy_, vie_encoder_.get(),
       module_process_thread, call_stats, congestion_controller, packet_router,
       bitrate_allocator, send_delay_stats, remb, event_log, &config_,
       encoder_config.max_bitrate_bps, suspended_ssrcs)));
 
   // Wait for ConstructionTask to complete so that |send_stream_| can be used.
   // |module_process_thread| must be registered and deregistered on the thread
   // it was created on.
   thread_sync_event_.Wait(rtc::Event::kForever);
@@ skipping 693 matching lines @@
       std::min(config_->rtp.max_packet_size,
                kPathMTU - transport_overhead_bytes_per_packet_);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     rtp_rtcp->SetMaxRtpPacketSize(rtp_packet_size);
   }
 }
 
 }  // namespace internal
 }  // namespace webrtc