Make LoadObserver settable per video send stream. Gives client flexibility and makes the implementa…

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 83 matching lines @@
   ss << ", render_delay_ms: " << render_delay_ms;
   ss << ", target_delay_ms: " << target_delay_ms;
   ss << ", suspend_below_min_bitrate: " << (suspend_below_min_bitrate ? "on"
                                                                       : "off");
   ss << '}';
   return ss.str();
 }
 
 namespace internal {
 VideoSendStream::VideoSendStream(
-    CpuOveruseObserver* overuse_observer,
     int num_cpu_cores,
     ProcessThread* module_process_thread,
     ChannelGroup* channel_group,
     int channel_id,
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
     const std::map<uint32_t, RtpState>& suspended_ssrcs)
     : transport_adapter_(config.send_transport),
       encoded_frame_proxy_(config.post_encode_callback),
       config_(config),
@@ skipping 40 matching lines @@
                                   config_.rtp.fec.ulpfec_payload_type);
   vie_encoder_->UpdateProtectionMethod(enable_protection_nack,
                                        enable_protection_fec);
 
   ConfigureSsrcs();
 
   vie_channel_->SetRTCPCName(config_.rtp.c_name.c_str());
 
   input_.reset(new internal::VideoCaptureInput(
       module_process_thread_, vie_encoder_, config_.local_renderer,
-      &stats_proxy_, overuse_observer));
+      &stats_proxy_, this));
 
   // 28 to match packet overhead in ModuleRtpRtcpImpl.
   DCHECK_LE(config_.rtp.max_packet_size, static_cast<size_t>(0xFFFF - 28));
   vie_channel_->SetMTU(static_cast<uint16_t>(config_.rtp.max_packet_size + 28));
 
   DCHECK(config.encoder_settings.encoder != nullptr);
   DCHECK_GE(config.encoder_settings.payload_type, 0);
   DCHECK_LE(config.encoder_settings.payload_type, 127);
   // TODO(pbos): Wire up codec internal-source setting or remove setting.
   CHECK_EQ(0, vie_encoder_->RegisterExternalEncoder(
@@ skipping 207 matching lines @@
 }
 
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
 }
 
 VideoSendStream::Stats VideoSendStream::GetStats() {
   return stats_proxy_.GetStats();
 }
 
+void VideoSendStream::OveruseDetected() {
+  if (config_.overuse_callback) {

[pbos-webrtc, 2015/09/03 20:44:55]

Remove {}s for consistency outside of talk/.

[the sun, 2015/09/03 20:54:55]

lol

[pthatcher1, 2015/09/04 04:56:03]

Please keep {}s to be consistent with code inside of webrtc/p2p, webrtc/base,
MediaController, and WebRtcVidoeEngine2

+    config_.overuse_callback->OnLoadUpdate(LoadObserver::kOveruse);
+  }
+}
+
+void VideoSendStream::NormalUsage() {
+  if (config_.overuse_callback) {

[pbos-webrtc, 2015/09/03 20:44:55]

Remove {}s for consistency outside of talk/.

[the sun, 2015/09/03 20:54:55]

lol

[pthatcher1, 2015/09/04 04:56:03]

Please keep {}s to be consistent with code inside of webrtc/p2p, webrtc/base,
MediaController, and WebRtcVidoeEngine2

+    config_.overuse_callback->OnLoadUpdate(LoadObserver::kUnderuse);
+  }
+}
+
 void VideoSendStream::ConfigureSsrcs() {
   vie_channel_->SetSSRC(config_.rtp.ssrcs.front(), kViEStreamTypeNormal, 0);
   for (size_t i = 0; i < config_.rtp.ssrcs.size(); ++i) {
     uint32_t ssrc = config_.rtp.ssrcs[i];
     vie_channel_->SetSSRC(ssrc, kViEStreamTypeNormal,
                           static_cast<unsigned char>(i));
     RtpStateMap::iterator it = suspended_ssrcs_.find(ssrc);
     if (it != suspended_ssrcs_.end())
       vie_channel_->SetRtpStateForSsrc(ssrc, it->second);
   }
@@ skipping 98 matching lines @@
                                        vie_channel_->IsSendingFecEnabled());
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc