Add separate send-side UMA stats for screenshare and video.

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 99 matching lines @@
 namespace internal {
 VideoSendStream::VideoSendStream(
     int num_cpu_cores,
     ProcessThread* module_process_thread,
     CallStats* call_stats,
     CongestionController* congestion_controller,
     BitrateAllocator* bitrate_allocator,
     const VideoSendStream::Config& config,
     const VideoEncoderConfig& encoder_config,
     const std::map<uint32_t, RtpState>& suspended_ssrcs)
-    : stats_proxy_(Clock::GetRealTimeClock(), config),
+    : 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),
       encoder_feedback_(new EncoderStateFeedback()),
       use_config_bitrate_(true) {
   LOG(LS_INFO) << "VideoSendStream: " << config_.ToString();
@@ skipping 297 matching lines @@
 
   if (!SetSendCodec(video_codec))
     return false;
 
   // Clear stats for disabled layers.
   for (size_t i = video_codec.numberOfSimulcastStreams;
        i < config_.rtp.ssrcs.size(); ++i) {
     stats_proxy_.OnInactiveSsrc(config_.rtp.ssrcs[i]);
   }
 
+  stats_proxy_.SetContentType(config.content_type);
+
   RTC_DCHECK_GE(config.min_transmit_bitrate_bps, 0);
   vie_encoder_->SetMinTransmitBitrate(config.min_transmit_bitrate_bps / 1000);
 
   encoder_config_ = config;
   use_config_bitrate_ = false;
   return true;
 }
 
 bool VideoSendStream::DeliverRtcp(const uint8_t* packet, size_t length) {
   return vie_channel_->ReceivedRTCPPacket(packet, length) == 0;
@@ skipping 123 matching lines @@
   vie_encoder_->SetSsrcs(used_ssrcs);
 
   // Restart the media flow
   vie_encoder_->Restart();
 
   return true;
 }
 
 }  // namespace internal
 }  // namespace webrtc