Define RtpTransportControllerSendInterface.

webrtc/voice_engine/channel.cc

 /*
  *  Copyright (c) 2012 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/voice_engine/channel.h"
 
 #include <algorithm>
 #include <utility>
 
 #include "webrtc/audio/utility/audio_frame_operations.h"
 #include "webrtc/base/array_view.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/location.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_limiter.h"
 #include "webrtc/base/timeutils.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/audio_coding/codecs/audio_format_conversion.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 #include "webrtc/modules/audio_processing/include/audio_processing.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/rtp_rtcp/include/receive_statistics.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_payload_registry.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
@@ skipping 2303 matching lines @@
 
 void Channel::EnableReceiveTransportSequenceNumber(int id) {
   rtp_header_parser_->DeregisterRtpHeaderExtension(
       kRtpExtensionTransportSequenceNumber);
   bool ret = rtp_header_parser_->RegisterRtpHeaderExtension(
       kRtpExtensionTransportSequenceNumber, id);
   RTC_DCHECK(ret);
 }
 
 void Channel::RegisterSenderCongestionControlObjects(
-    RtpPacketSender* rtp_packet_sender,
-    TransportFeedbackObserver* transport_feedback_observer,
-    PacketRouter* packet_router,
+    RtpTransportControllerSendInterface* transport,
     RtcpBandwidthObserver* bandwidth_observer) {
+  RtpPacketSender* rtp_packet_sender = transport->packet_sender();
+  TransportFeedbackObserver* transport_feedback_observer =
+      transport->transport_feedback_observer();
+  PacketRouter* packet_router = transport->packet_router();
+
   RTC_DCHECK(rtp_packet_sender);
   RTC_DCHECK(transport_feedback_observer);
   RTC_DCHECK(packet_router && !packet_router_);
   rtcp_observer_->SetBandwidthObserver(bandwidth_observer);
   feedback_observer_proxy_->SetTransportFeedbackObserver(
       transport_feedback_observer);
   seq_num_allocator_proxy_->SetSequenceNumberAllocator(packet_router);
   rtp_packet_sender_proxy_->SetPacketSender(rtp_packet_sender);
   _rtpRtcpModule->SetStorePacketsStatus(true, 600);
   packet_router->AddRtpModule(_rtpRtcpModule.get());
@@ skipping 641 matching lines @@
   int64_t min_rtt = 0;
   if (_rtpRtcpModule->RTT(remoteSSRC, &rtt, &avg_rtt, &min_rtt, &max_rtt) !=
       0) {
     return 0;
   }
   return rtt;
 }
 
 }  // namespace voe
 }  // namespace webrtc