Add NACK rate throttling for audio channels.

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.
  */
 #include "webrtc/base/checks.h"
 #include "webrtc/common.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/codecs/builtin_audio_decoder_factory.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
+#include "webrtc/voice_engine/voice_engine_impl.h"
+#include "webrtc/voice_engine/channel_proxy.h"
 
 namespace webrtc {
 namespace test {
 
 namespace {
 const int kVideoRotationRtpExtensionId = 4;
 }
 
 CallTest::CallTest()
     : clock_(Clock::GetRealTimeClock()),
       video_send_config_(nullptr),
       video_send_stream_(nullptr),
       audio_send_config_(nullptr),
       audio_send_stream_(nullptr),
+      audio_send_channel_proxy_(nullptr),
       fake_encoder_(clock_),
       num_video_streams_(1),
       num_audio_streams_(0),
       decoder_factory_(CreateBuiltinAudioDecoderFactory()),
       fake_send_audio_device_(nullptr),
       fake_recv_audio_device_(nullptr) {}
 
 CallTest::~CallTest() {
 }
 
@@ skipping 58 matching lines @@
   }
 
   Start();
   test->PerformTest();
   send_transport_->StopSending();
   receive_transport_->StopSending();
   Stop();
 
   DestroyStreams();
   DestroyCalls();
-  if (num_audio_streams_ > 0)
+  if (num_audio_streams_ > 0) {
+    audio_send_channel_proxy_.reset();
     DestroyVoiceEngines();
+  }
 }
 
 void CallTest::Start() {
   if (video_send_stream_)
     video_send_stream_->Start();
   for (VideoReceiveStream* video_recv_stream : video_receive_streams_)
     video_recv_stream->Start();
   if (audio_send_stream_) {
     fake_send_audio_device_->Start();
     audio_send_stream_->Start();
@@ skipping 187 matching lines @@
   voe_send_.voice_engine = VoiceEngine::Create();
   voe_send_.base = VoEBase::GetInterface(voe_send_.voice_engine);
   voe_send_.codec = VoECodec::GetInterface(voe_send_.voice_engine);
   EXPECT_EQ(0, voe_send_.base->Init(fake_send_audio_device_.get(), nullptr,
                                     decoder_factory_));
   Config voe_config;
   voe_config.Set<VoicePacing>(new VoicePacing(true));
   voe_send_.channel_id = voe_send_.base->CreateChannel(voe_config);
   EXPECT_GE(voe_send_.channel_id, 0);
 
+  audio_send_channel_proxy_ =
+      static_cast<VoiceEngineImpl*>(voe_send_.voice_engine)
+          ->GetChannelProxy(voe_send_.channel_id);
+
   voe_recv_.voice_engine = VoiceEngine::Create();
   voe_recv_.base = VoEBase::GetInterface(voe_recv_.voice_engine);
   voe_recv_.codec = VoECodec::GetInterface(voe_recv_.voice_engine);
   EXPECT_EQ(0, voe_recv_.base->Init(fake_recv_audio_device_.get(), nullptr,
                                     decoder_factory_));
   voe_recv_.channel_id = voe_recv_.base->CreateChannel();
   EXPECT_GE(voe_recv_.channel_id, 0);
 }
 
 void CallTest::DestroyVoiceEngines() {
@@ skipping 100 matching lines @@
 
 EndToEndTest::EndToEndTest(unsigned int timeout_ms) : BaseTest(timeout_ms) {
 }
 
 bool EndToEndTest::ShouldCreateReceivers() const {
   return true;
 }
 
 }  // namespace test
 }  // namespace webrtc