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

webrtc/video/send_statistics_proxy.h

 /*
  *  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 20 matching lines @@
                             public RtcpStatisticsCallback,
                             public RtcpPacketTypeCounterObserver,
                             public StreamDataCountersCallback,
                             public BitrateStatisticsObserver,
                             public FrameCountObserver,
                             public VideoEncoderRateObserver,
                             public SendSideDelayObserver {
  public:
   static const int kStatsTimeoutMs;
 
-  SendStatisticsProxy(Clock* clock, const VideoSendStream::Config& config);
+  SendStatisticsProxy(Clock* clock,
+                      const VideoSendStream::Config& config,
+                      const VideoEncoderConfig& encoder_config);
   virtual ~SendStatisticsProxy();
 
   VideoSendStream::Stats GetStats();
 
   virtual void OnSendEncodedImage(const EncodedImage& encoded_image,
                                   const RTPVideoHeader* rtp_video_header);
   // Used to update incoming frame rate.
   void OnIncomingFrame(int width, int height);
 
   // Used to update encode time of frames.
@@ skipping 53 matching lines @@
     void Add(bool sample);
     int Percent(int min_required_samples) const;
     int Permille(int min_required_samples) const;
 
    private:
     int Fraction(int min_required_samples, float multiplier) const;
     int sum;
     int num_samples;
   };
   struct StatsUpdateTimes {
-    StatsUpdateTimes() : resolution_update_ms(0) {}
+    StatsUpdateTimes() : resolution_update_ms(0), bitrate_update_ms(0) {}
     int64_t resolution_update_ms;
     int64_t bitrate_update_ms;
   };
   void PurgeOldStats() EXCLUSIVE_LOCKS_REQUIRED(crit_);
   VideoSendStream::StreamStats* GetStatsEntry(uint32_t ssrc)
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
   void UpdateHistograms() EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   Clock* const clock_;
   const VideoSendStream::Config config_;
   mutable rtc::CriticalSection crit_;
   VideoSendStream::Stats stats_ GUARDED_BY(crit_);
+  const std::string uma_prefix_;
   rtc::RateTracker input_frame_rate_tracker_ GUARDED_BY(crit_);
   rtc::RateTracker sent_frame_rate_tracker_ GUARDED_BY(crit_);
   uint32_t last_sent_frame_timestamp_ GUARDED_BY(crit_);
   std::map<uint32_t, StatsUpdateTimes> update_times_ GUARDED_BY(crit_);
 
   int max_sent_width_per_timestamp_ GUARDED_BY(crit_);
   int max_sent_height_per_timestamp_ GUARDED_BY(crit_);
   SampleCounter input_width_counter_ GUARDED_BY(crit_);
   SampleCounter input_height_counter_ GUARDED_BY(crit_);
   SampleCounter sent_width_counter_ GUARDED_BY(crit_);
   SampleCounter sent_height_counter_ GUARDED_BY(crit_);
   SampleCounter encode_time_counter_ GUARDED_BY(crit_);
   BoolSampleCounter key_frame_counter_ GUARDED_BY(crit_);
   BoolSampleCounter quality_limited_frame_counter_ GUARDED_BY(crit_);
   SampleCounter quality_downscales_counter_ GUARDED_BY(crit_);
   BoolSampleCounter bw_limited_frame_counter_ GUARDED_BY(crit_);
   SampleCounter bw_resolutions_disabled_counter_ GUARDED_BY(crit_);
   SampleCounter delay_counter_ GUARDED_BY(crit_);
   SampleCounter max_delay_counter_ GUARDED_BY(crit_);
 };
 
 }  // namespace webrtc
 #endif  // WEBRTC_VIDEO_SEND_STATISTICS_PROXY_H_