Move RtpTransportControllerSend to a new file.

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.
  */
 
@@ skipping 20 matching lines @@
 #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"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/receive_side_congestion_controller.h"
-#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h"
 #include "webrtc/modules/pacing/paced_sender.h"
 #include "webrtc/modules/rtp_rtcp/include/flexfec_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_header_extension.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_received.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/system_wrappers/include/cpu_info.h"
 #include "webrtc/system_wrappers/include/metrics.h"
@@ skipping 28 matching lines @@
 }
 
 bool UseSendSideBwe(const AudioReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp.extensions, config.rtp.transport_cc);
 }
 
 bool UseSendSideBwe(const FlexfecReceiveStream::Config& config) {
   return UseSendSideBwe(config.rtp_header_extensions, config.transport_cc);
 }
 
-class RtpTransportControllerSend : public RtpTransportControllerSendInterface {
- public:
-  RtpTransportControllerSend(Clock* clock, webrtc::RtcEventLog* event_log);
-
-  void RegisterNetworkObserver(
-      SendSideCongestionController::Observer* observer);
-
-  // Implements RtpTransportControllerSendInterface
-  PacketRouter* packet_router() override { return &packet_router_; }
-  SendSideCongestionController* send_side_cc() override {
-    return &send_side_cc_;
-  }
-  TransportFeedbackObserver* transport_feedback_observer() override {
-    return &send_side_cc_;
-  }
-  RtpPacketSender* packet_sender() override { return send_side_cc_.pacer(); }
-
- private:
-  PacketRouter packet_router_;
-  SendSideCongestionController send_side_cc_;
-};
-
-RtpTransportControllerSend::RtpTransportControllerSend(
-    Clock* clock,
-    webrtc::RtcEventLog* event_log)
-    : send_side_cc_(clock, nullptr /* observer */, event_log, &packet_router_) {
-}
-
-void RtpTransportControllerSend::RegisterNetworkObserver(
-    SendSideCongestionController::Observer* observer) {
-  // Must be called only once.
-  send_side_cc_.RegisterNetworkObserver(observer);
-}
-
 }  // namespace
 
 namespace internal {
 
 class Call : public webrtc::Call,
              public PacketReceiver,
              public RecoveredPacketReceiver,
              public SendSideCongestionController::Observer,
              public BitrateAllocator::LimitObserver {
  public:
@@ skipping 1183 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