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

webrtc/test/call_test.cc

 /*
  *  Copyright (c) 2014 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 224 matching lines @@
   RTC_DCHECK(num_audio_streams == 0 || voe_send_.channel_id >= 0);
   if (num_video_streams > 0) {
     video_send_config_ = VideoSendStream::Config(send_transport);
     video_send_config_.encoder_settings.encoder = &fake_encoder_;
     video_send_config_.encoder_settings.payload_name = "FAKE";
     video_send_config_.encoder_settings.payload_type =
         kFakeVideoSendPayloadType;
     video_send_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTransportSequenceNumberUri,
                      kTransportSequenceNumberExtensionId));
+    video_send_config_.rtp.extensions.push_back(RtpExtension(
+        RtpExtension::kVideoContentTypeUri, kVideoContentTypeExtensionId));
     FillEncoderConfiguration(num_video_streams, &video_encoder_config_);
 
     for (size_t i = 0; i < num_video_streams; ++i)
       video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[i]);
     video_send_config_.rtp.extensions.push_back(RtpExtension(
         RtpExtension::kVideoRotationUri, kVideoRotationRtpExtensionId));
   }
 
   if (num_audio_streams > 0) {
     audio_send_config_ = AudioSendStream::Config(send_transport);
@@ skipping 319 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc