Inform jitter buffer about FlexFEC protection.

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 82 matching lines @@
     test->ModifyVideoConfigs(&video_send_config_, &video_receive_configs_,
                              &video_encoder_config_);
   }
   if (num_audio_streams_ > 0) {
     test->ModifyAudioConfigs(&audio_send_config_, &audio_receive_configs_);
   }
   if (num_flexfec_streams_ > 0) {
     test->ModifyFlexfecConfigs(&flexfec_receive_configs_);
   }
 
+  if (num_flexfec_streams_ > 0) {
+    CreateFlexfecStreams();
+    test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
+  }
   if (num_video_streams_ > 0) {
     CreateVideoStreams();
     test->OnVideoStreamsCreated(video_send_stream_, video_receive_streams_);
   }
   if (num_audio_streams_ > 0) {
     CreateAudioStreams();
     test->OnAudioStreamsCreated(audio_send_stream_, audio_receive_streams_);
   }
-  if (num_flexfec_streams_ > 0) {
-    CreateFlexfecStreams();
-    test->OnFlexfecStreamsCreated(flexfec_receive_streams_);
-  }
 
   if (num_video_streams_ > 0) {
     int width = kDefaultWidth;
     int height = kDefaultHeight;
     int frame_rate = kDefaultFramerate;
     test->ModifyVideoCaptureStartResolution(&width, &height, &frame_rate);
     CreateFrameGeneratorCapturer(frame_rate, width, height);
     test->OnFrameGeneratorCapturerCreated(frame_generator_capturer_.get());
   }
 
@@ skipping 220 matching lines @@
 
 void CallTest::CreateFlexfecStreams() {
   for (size_t i = 0; i < flexfec_receive_configs_.size(); ++i) {
     flexfec_receive_streams_.push_back(
         receiver_call_->CreateFlexfecReceiveStream(
             flexfec_receive_configs_[i]));
   }
 }
 
 void CallTest::DestroyStreams() {
+  if (audio_send_stream_)
+    sender_call_->DestroyAudioSendStream(audio_send_stream_);
+  audio_send_stream_ = nullptr;
+  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
+    receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
+
   if (video_send_stream_)
     sender_call_->DestroyVideoSendStream(video_send_stream_);
   video_send_stream_ = nullptr;
   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
     receiver_call_->DestroyVideoReceiveStream(video_recv_stream);
 
-  if (audio_send_stream_)
-    sender_call_->DestroyAudioSendStream(audio_send_stream_);
-  audio_send_stream_ = nullptr;
-  for (AudioReceiveStream* audio_recv_stream : audio_receive_streams_)
-    receiver_call_->DestroyAudioReceiveStream(audio_recv_stream);
-
   for (FlexfecReceiveStream* flexfec_recv_stream : flexfec_receive_streams_)
     receiver_call_->DestroyFlexfecReceiveStream(flexfec_recv_stream);
 
   video_receive_streams_.clear();
   allocated_decoders_.clear();
 }
 
 void CallTest::CreateVoiceEngines() {
   CreateFakeAudioDevices();
   voe_send_.voice_engine = VoiceEngine::Create();
@@ skipping 132 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc