Remove unused VideoSendStream TransportAdapter.

webrtc/video/video_send_stream.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 12 matching lines @@
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/pacing/packet_router.h"
 #include "webrtc/modules/utility/include/process_thread.h"
 #include "webrtc/video/call_stats.h"
 #include "webrtc/video/video_capture_input.h"
 #include "webrtc/video/vie_remb.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 
-class PacedSender;
 class RtcpIntraFrameObserver;
 class TransportFeedbackObserver;
 
 std::string
 VideoSendStream::Config::EncoderSettings::ToString() const {
   std::stringstream ss;
   ss << "{payload_name: " << payload_name;
   ss << ", payload_type: " << payload_type;
   ss << ", encoder: " << (encoder != nullptr ? "(VideoEncoder)" : "nullptr");
   ss << '}';
@@ skipping 109 matching lines @@
     CallStats* call_stats,
     CongestionController* congestion_controller,
     VieRemb* remb,
     BitrateAllocator* bitrate_allocator,
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
     const std::map<uint32_t, RtpState>& suspended_ssrcs)
     : stats_proxy_(Clock::GetRealTimeClock(),
                    config,
                    encoder_config.content_type),
-      transport_adapter_(config.send_transport),
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
       suspended_ssrcs_(suspended_ssrcs),
       module_process_thread_(module_process_thread),
       call_stats_(call_stats),
       congestion_controller_(congestion_controller),
       remb_(remb),
       overuse_detector_(
           Clock::GetRealTimeClock(),
           GetCpuOveruseOptions(config.encoder_settings.full_overuse_time),
@@ skipping 153 matching lines @@
 
   congestion_controller_->GetRemoteBitrateEstimator(false)->RemoveStream(
       vie_receiver_->GetRemoteSsrc());
 }
 
 VideoCaptureInput* VideoSendStream::Input() {
   return &input_;
 }
 
 void VideoSendStream::Start() {
-  transport_adapter_.Enable();
   vie_encoder_.Pause();
   if (vie_channel_.StartSend() == 0) {
     // Was not already started, trigger a keyframe.
     vie_encoder_.SendKeyFrame();
   }
   vie_encoder_.Restart();
   vie_receiver_->StartReceive();
 }
 
 void VideoSendStream::Stop() {
   // TODO(pbos): Make sure the encoder stops here.
   vie_channel_.StopSend();
   vie_receiver_->StopReceive();
-  transport_adapter_.Disable();
 }
 
 bool VideoSendStream::ReconfigureVideoEncoder(
     const VideoEncoderConfig& config) {
   TRACE_EVENT0("webrtc", "VideoSendStream::(Re)configureVideoEncoder");
   LOG(LS_INFO) << "(Re)configureVideoEncoder: " << config.ToString();
   const std::vector<VideoStream>& streams = config.streams;
   RTC_DCHECK(!streams.empty());
   RTC_DCHECK_GE(config_.rtp.ssrcs.size(), streams.size());
 
@@ skipping 266 matching lines @@
   used_ssrcs.resize(static_cast<size_t>(video_codec.numberOfSimulcastStreams));
   vie_encoder_.SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_.Restart();
 
   return true;
 }
 }  // namespace internal
 }  // namespace webrtc