Simple tests for Call::SetBitrateConfig.

webrtc/audio/audio_send_stream_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.
  */
 
@@ skipping 116 matching lines @@
 
 struct ConfigHelper {
   ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call)
       : stream_config_(nullptr),
         simulated_clock_(123456),
         send_side_cc_(rtc::MakeUnique<SendSideCongestionController>(
             &simulated_clock_,
             nullptr /* observer */,
             &event_log_,
             &packet_router_)),
-        fake_transport_(send_side_cc_.get()),
+        fake_transport_(&packet_router_, send_side_cc_.get()),
         bitrate_allocator_(&limit_observer_),
         worker_queue_("ConfigHelper_worker_queue") {
     using testing::Invoke;
 
     EXPECT_CALL(voice_engine_,
         RegisterVoiceEngineObserver(_)).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_,
         DeRegisterVoiceEngineObserver()).WillOnce(Return(0));
     EXPECT_CALL(voice_engine_, audio_device_module());
     EXPECT_CALL(voice_engine_, audio_processing());
@@ skipping 389 matching lines @@
   stream_config.send_codec_spec->cng_payload_type = rtc::Optional<int>(105);
   internal::AudioSendStream send_stream(
       stream_config, helper.audio_state(), helper.worker_queue(),
       helper.transport(), helper.bitrate_allocator(), helper.event_log(),
       helper.rtcp_rtt_stats());
   send_stream.Reconfigure(stream_config);
 }
 
 }  // namespace test
 }  // namespace webrtc