Implement qpSum stat for video send ssrc stats.

webrtc/video_send_stream.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 39 matching lines @@
   };
 
   struct Stats {
     std::string ToString(int64_t time_ms) const;
     std::string encoder_implementation_name = "unknown";
     int input_frame_rate = 0;
     int encode_frame_rate = 0;
     int avg_encode_time_ms = 0;
     int encode_usage_percent = 0;
     uint32_t frames_encoded = 0;
+    rtc::Optional<uint64_t> qp_sum;
     // Bitrate the encoder is currently configured to use due to bandwidth
     // limitations.
     int target_media_bitrate_bps = 0;
     // Bitrate the encoder is actually producing.
     int media_bitrate_bps = 0;
     // Media bitrate this VideoSendStream is configured to prefer if there are
     // no bandwidth limitations.
     int preferred_media_bitrate_bps = 0;
     bool suspended = false;
     bool bw_limited_resolution = false;
@@ skipping 148 matching lines @@
     EnableEncodedFrameRecording(std::vector<rtc::PlatformFile>(), 0);
   }
 
  protected:
   virtual ~VideoSendStream() {}
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_VIDEO_SEND_STREAM_H_