Add NACK rate throttling for audio channels.

webrtc/test/call_test.h

 /*
  *  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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
 #define WEBRTC_TEST_CALL_TEST_H_
 
 #include <memory>
 #include <vector>
 
 #include "webrtc/call.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 
 namespace webrtc {
 
 class VoEBase;
 class VoECodec;
 
+namespace voe {
+class ChannelProxy;
+}
+
 namespace test {
 
 class BaseTest;
 
 class CallTest : public ::testing::Test {
  public:
   CallTest();
   virtual ~CallTest();
 
   static const size_t kNumSsrcs = 3;
@@ skipping 44 matching lines @@
 
   Clock* const clock_;
 
   std::unique_ptr<Call> sender_call_;
   std::unique_ptr<PacketTransport> send_transport_;
   VideoSendStream::Config video_send_config_;
   VideoEncoderConfig video_encoder_config_;
   VideoSendStream* video_send_stream_;
   AudioSendStream::Config audio_send_config_;
   AudioSendStream* audio_send_stream_;
+  std::unique_ptr<voe::ChannelProxy> audio_send_channel_proxy_;
 
   std::unique_ptr<Call> receiver_call_;
   std::unique_ptr<PacketTransport> receive_transport_;
   std::vector<VideoReceiveStream::Config> video_receive_configs_;
   std::vector<VideoReceiveStream*> video_receive_streams_;
   std::vector<AudioReceiveStream::Config> audio_receive_configs_;
   std::vector<AudioReceiveStream*> audio_receive_streams_;
 
   std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
   test::FakeEncoder fake_encoder_;
@@ skipping 78 matching lines @@
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_