Add QP sum stats for received streams.

webrtc/video/receive_statistics_proxy.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 410 matching lines @@
   } else {
     auto it = rtx_stats_.find(ssrc);
     if (it != rtx_stats_.end()) {
       it->second = counters;
     } else {
       RTC_NOTREACHED() << "Unexpected stream ssrc: " << ssrc;
     }
   }
 }
 
-void ReceiveStatisticsProxy::OnDecodedFrame() {
+void ReceiveStatisticsProxy::OnDecodedFrame(rtc::Optional<uint8_t> qp) {
   uint64_t now = clock_->TimeInMilliseconds();
 
   rtc::CritScope lock(&crit_);
   ++stats_.frames_decoded;
+  if (qp) {
+    if (!stats_.qp_sum) {
+      if (stats_.frames_decoded != 1) {
+        LOG(LS_WARNING)
+            << "Frames decoded was not 1 when first qp value was received.";
+        stats_.frames_decoded = 1;
+      }
+      stats_.qp_sum = rtc::Optional<uint64_t>(0);
+    }
+    *stats_.qp_sum += *qp;
+  } else if (stats_.qp_sum) {
+    LOG(LS_WARNING)
+        << "QP sum was already set and no QP was given for a frame.";
+    stats_.qp_sum = rtc::Optional<uint64_t>();
+  }
   decode_fps_estimator_.Update(1, now);
   stats_.decode_frame_rate = decode_fps_estimator_.Rate(now).value_or(0);
 }
 
 void ReceiveStatisticsProxy::OnRenderedFrame(const VideoFrame& frame) {
   int width = frame.width();
   int height = frame.height();
   RTC_DCHECK_GT(width, 0);
   RTC_DCHECK_GT(height, 0);
   uint64_t now = clock_->TimeInMilliseconds();
@@ skipping 90 matching lines @@
   sum = 0;
 }
 
 void ReceiveStatisticsProxy::OnRttUpdate(int64_t avg_rtt_ms,
                                          int64_t max_rtt_ms) {
   rtc::CritScope lock(&crit_);
   avg_rtt_ms_ = avg_rtt_ms;
 }
 
 }  // namespace webrtc