Allow mocking SendSideCongestionController for Call tests.

webrtc/call/call_unittest.cc

 /*
  *  Copyright (c) 2015 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 <list>
 #include <map>
 #include <memory>
+#include <utility>
 
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/call/audio_state.h"
 #include "webrtc/call/call.h"
+#include "webrtc/call/fake_rtp_transport_controller_send.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/audio_mixer/audio_mixer_impl.h"
+#include "webrtc/modules/congestion_controller/include/mock/mock_send_side_congestion_controller.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/mock_audio_decoder_factory.h"
 #include "webrtc/test/mock_transport.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace {
 
 struct CallHelper {
   explicit CallHelper(
       rtc::scoped_refptr<webrtc::AudioDecoderFactory> decoder_factory = nullptr)
@@ skipping 269 matching lines @@
   config.remote_ssrc = 5548;
   stream = call->CreateFlexfecReceiveStream(config);
   EXPECT_NE(stream, nullptr);
   streams.push_back(stream);
 
   for (auto s : streams) {
     call->DestroyFlexfecReceiveStream(s);
   }
 }
 
+// TODO(zstein): This is just a motivating example for
+// MockSendSideCongestionController. It should be deleted once we have more
+// meaningful tests.
+TEST(CallTest, MockSendSideCongestionControllerExample) {
+  RtcEventLogNullImpl event_log;
+  Call::Config config(&event_log);
+
+  SimulatedClock clock(123456);
+  PacketRouter packet_router;
+  testing::NiceMock<test::MockSendSideCongestionController> mock_cc(
+      &clock, &event_log, &packet_router);
+  auto transport_send = rtc::MakeUnique<FakeRtpTransportController>(&mock_cc);
+  std::unique_ptr<Call> call(Call::Create(config, std::move(transport_send)));
+
+  Call::Config::BitrateConfig bitrate_config;
+  bitrate_config.min_bitrate_bps = 1;
+  bitrate_config.start_bitrate_bps = 2;
+  bitrate_config.max_bitrate_bps = 3;
+
+  EXPECT_CALL(mock_cc, SetBweBitrates(1, 2, 3));
+  call->SetBitrateConfig(bitrate_config);
+}
+
 }  // namespace webrtc