Allow mocking SendSideCongestionController for Call tests.

webrtc/call/call.cc

 /*
  *  Copyright (c) 2013 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 <string.h>
 #include <algorithm>
 #include <map>
 #include <memory>
 #include <set>
 #include <utility>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
 #include "webrtc/audio/audio_send_stream.h"
 #include "webrtc/audio/audio_state.h"
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/basictypes.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/location.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/ptr_util.h"
 #include "webrtc/base/task_queue.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/call/bitrate_allocator.h"
 #include "webrtc/call/call.h"
 #include "webrtc/call/flexfec_receive_stream_impl.h"
 #include "webrtc/call/rtp_transport_controller_send.h"
 #include "webrtc/config.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
@@ skipping 51 matching lines @@
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
   Call(const Call::Config& config,
-       std::unique_ptr<RtpTransportControllerSend> transport_send);
+       std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
   virtual ~Call();
 
   // Implements webrtc::Call.
   PacketReceiver* Receiver() override;
 
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
 
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
@@ skipping 180 matching lines @@
   ss << "send_bw_bps: " << send_bandwidth_bps << ", ";
   ss << "recv_bw_bps: " << recv_bandwidth_bps << ", ";
   ss << "max_pad_bps: " << max_padding_bitrate_bps << ", ";
   ss << "pacer_delay_ms: " << pacer_delay_ms << ", ";
   ss << "rtt_ms: " << rtt_ms;
   ss << '}';
   return ss.str();
 }
 
 Call* Call::Create(const Call::Config& config) {
-  return new internal::Call(
-      config, std::unique_ptr<RtpTransportControllerSend>(
-                  new RtpTransportControllerSend(Clock::GetRealTimeClock(),
-                                                 config.event_log)));
+  return new internal::Call(config,
+                            rtc::MakeUnique<RtpTransportControllerSend>(
+                                Clock::GetRealTimeClock(), config.event_log));
+}
+
+Call* Call::Create(
+    const Call::Config& config,
+    std::unique_ptr<RtpTransportControllerSendInterface> transport_send) {
+  return new internal::Call(config, std::move(transport_send));
 }
 
 namespace internal {
 
 Call::Call(const Call::Config& config,
-           std::unique_ptr<RtpTransportControllerSend> transport_send)
+           std::unique_ptr<RtpTransportControllerSendInterface> transport_send)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
       module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
       pacer_thread_(ProcessThread::Create("PacerThread")),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator(this)),
       config_(config),
       audio_network_state_(kNetworkDown),
       video_network_state_(kNetworkDown),
       receive_crit_(RWLockWrapper::CreateRWLock()),
@@ skipping 14 matching lines @@
   RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
   RTC_DCHECK(config.event_log != nullptr);
   RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
   RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
                 config.bitrate_config.min_bitrate_bps);
   if (config.bitrate_config.max_bitrate_bps != -1) {
     RTC_DCHECK_GE(config.bitrate_config.max_bitrate_bps,
                   config.bitrate_config.start_bitrate_bps);
   }
   Trace::CreateTrace();
-  transport_send->RegisterNetworkObserver(this);
+  transport_send->send_side_cc()->RegisterNetworkObserver(this);
   transport_send_ = std::move(transport_send);
   transport_send_->send_side_cc()->SignalNetworkState(kNetworkDown);
   transport_send_->send_side_cc()->SetBweBitrates(
       config_.bitrate_config.min_bitrate_bps,
       config_.bitrate_config.start_bitrate_bps,
       config_.bitrate_config.max_bitrate_bps);
   call_stats_->RegisterStatsObserver(&receive_side_cc_);
   call_stats_->RegisterStatsObserver(transport_send_->send_side_cc());
 
   module_process_thread_->Start();
@@ skipping 926 matching lines @@
       (use_send_side_bwe && header.extension.hasTransportSequenceNumber)) {
     receive_side_cc_.OnReceivedPacket(
         packet.arrival_time_ms(), packet.payload_size() + packet.padding_size(),
         header);
   }
 }
 
 }  // namespace internal
 
 }  // namespace webrtc