Remove webrtc/stream.h and unutilized inheritance.

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 473 matching lines @@
   rtp_rtcp_modules_[0]->SetREMBStatus(false);
   remb_->RemoveRembSender(rtp_rtcp_modules_[0]);
 
   for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_) {
     congestion_controller_->packet_router()->RemoveRtpModule(rtp_rtcp);
     module_process_thread_->DeRegisterModule(rtp_rtcp);
     delete rtp_rtcp;
   }
 }
 
-VideoCaptureInput* VideoSendStream::Input() {
-  return &input_;
+void VideoSendStream::SignalNetworkState(NetworkState state) {
+  // When network goes up, enable RTCP status before setting transmission state.
+  // When it goes down, disable RTCP afterwards. This ensures that any packets
+  // sent due to the network state changed will not be dropped.
+  if (state == kNetworkUp) {
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+      rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode);
+  }
+  vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
+  if (state == kNetworkDown) {
+    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+      rtp_rtcp->SetRTCPStatus(RtcpMode::kOff);
+  }
+}
+
+bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
+  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
+    rtp_rtcp->IncomingRtcpPacket(packet, length);
+  return true;
 }
 
 void VideoSendStream::Start() {
   if (payload_router_.active())
     return;
   TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Start");
   vie_encoder_.Pause();
   payload_router_.set_active(true);
   // Was not already started, trigger a keyframe.
   vie_encoder_.SendKeyFrame();
   vie_encoder_.Restart();
 }
 
 void VideoSendStream::Stop() {
   if (!payload_router_.active())
     return;
   TRACE_EVENT_INSTANT0("webrtc", "VideoSendStream::Stop");
   // TODO(pbos): Make sure the encoder stops here.
   payload_router_.set_active(false);
 }
 
+VideoCaptureInput* VideoSendStream::Input() {
+  return &input_;
+}
+
 bool VideoSendStream::EncoderThreadFunction(void* obj) {
   static_cast<VideoSendStream*>(obj)->EncoderProcess();
   // We're done, return false to abort.
   return false;
 }
 
 void VideoSendStream::EncoderProcess() {
   RTC_CHECK_EQ(0, vie_encoder_.RegisterExternalEncoder(
                       config_.encoder_settings.encoder,
                       config_.encoder_settings.payload_type,
@@ skipping 46 matching lines @@
       config, config_.encoder_settings.payload_name,
       config_.encoder_settings.payload_type);
   {
     rtc::CritScope lock(&encoder_settings_crit_);
     pending_encoder_settings_ = rtc::Optional<EncoderSettings>(
         {video_codec, config.min_transmit_bitrate_bps});
   }
   encoder_wakeup_event_.Set();
 }
 
-bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
-  for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-    rtp_rtcp->IncomingRtcpPacket(packet, length);
-  return true;
-}
-
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
 void VideoSendStream::OveruseDetected() {
   if (config_.overuse_callback)
     config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
 }
 
 void VideoSendStream::NormalUsage() {
@@ skipping 108 matching lines @@
   }
 
   for (size_t i = 0; i < config_.rtp.rtx.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.rtx.ssrcs[i];
     rtp_states[ssrc] = rtp_rtcp_modules_[i]->GetRtxState();
   }
 
   return rtp_states;
 }
 
-void VideoSendStream::SignalNetworkState(NetworkState state) {
-  // When network goes up, enable RTCP status before setting transmission state.
-  // When it goes down, disable RTCP afterwards. This ensures that any packets
-  // sent due to the network state changed will not be dropped.
-  if (state == kNetworkUp) {
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-      rtp_rtcp->SetRTCPStatus(config_.rtp.rtcp_mode);
-  }
-  vie_encoder_.SetNetworkTransmissionState(state == kNetworkUp);
-  if (state == kNetworkDown) {
-    for (RtpRtcp* rtp_rtcp : rtp_rtcp_modules_)
-      rtp_rtcp->SetRTCPStatus(RtcpMode::kOff);
-  }
-}
-
 int VideoSendStream::GetPaddingNeededBps() const {
   return vie_encoder_.GetPaddingNeededBps();
 }
 
 void VideoSendStream::OnBitrateUpdated(uint32_t bitrate_bps,
                                        uint8_t fraction_loss,
                                        int64_t rtt) {
   vie_encoder_.OnBitrateUpdated(bitrate_bps, fraction_loss, rtt);
 }
 
@@ skipping 15 matching lines @@
                           &module_nack_rate);
     *sent_video_rate_bps += module_video_rate;
     *sent_nack_rate_bps += module_nack_rate;
     *sent_fec_rate_bps += module_fec_rate;
   }
   return 0;
 }
 
 }  // namespace internal
 }  // namespace webrtc