Rename incoming_framerate_ to capture_framerate_ in screenshare_layers.

webrtc/modules/video_coding/codecs/vp8/screenshare_layers.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.
 */
 
 #include "webrtc/modules/video_coding/codecs/vp8/screenshare_layers.h"
@@ skipping 137 matching lines @@
       flags = -1;
       ++stats_.num_dropped_frames_;
       break;
     default:
       flags = -1;
       RTC_NOTREACHED();
   }
 
   int64_t ts_diff;
   if (last_timestamp_ == -1) {
-    ts_diff =
-        kOneSecond90Khz / incoming_framerate_.value_or(*target_framerate_);
+    ts_diff = kOneSecond90Khz / capture_framerate_.value_or(*target_framerate_);
   } else {
     ts_diff = unwrapped_timestamp - last_timestamp_;
   }
   // Make sure both frame droppers leak out bits.
   layers_[0].UpdateDebt(ts_diff / 90);
   layers_[1].UpdateDebt(ts_diff / 90);
   last_timestamp_ = timestamp;
   return flags;
 }
 
 std::vector<uint32_t> ScreenshareLayers::OnRatesUpdated(int bitrate_kbps,
                                                         int max_bitrate_kbps,
                                                         int framerate) {
   RTC_DCHECK_GT(framerate, 0);
   if (!target_framerate_) {
     // First OnRatesUpdated() is called during construction, with the configured
     // targets as parameters.
     target_framerate_.emplace(framerate);
-    incoming_framerate_ = target_framerate_;
+    capture_framerate_ = target_framerate_;
     bitrate_updated_ = true;
   } else {
     bitrate_updated_ =
         bitrate_kbps != static_cast<int>(layers_[0].target_rate_kbps_) ||
         max_bitrate_kbps != static_cast<int>(layers_[1].target_rate_kbps_) ||
-        (incoming_framerate_ &&
-         framerate != static_cast<int>(*incoming_framerate_));
+        (capture_framerate_ &&
+         framerate != static_cast<int>(*capture_framerate_));
     if (framerate < 0) {
-      incoming_framerate_.reset();
+      capture_framerate_.reset();
     } else {
-      incoming_framerate_.emplace(framerate);
+      capture_framerate_.emplace(framerate);
     }
   }
 
   layers_[0].target_rate_kbps_ = bitrate_kbps;
   layers_[1].target_rate_kbps_ = max_bitrate_kbps;
 
   std::vector<uint32_t> allocation;
   allocation.push_back(bitrate_kbps);
   if (max_bitrate_kbps > bitrate_kbps)
     allocation.push_back(max_bitrate_kbps - bitrate_kbps);
@@ skipping 120 matching lines @@
       max_qp_ = cfg->rc_max_quantizer;
       // After a dropped frame, a frame with max qp will be encoded and the
       // quality will then ramp up from there. To boost the speed of recovery,
       // encode the next frame with lower max qp. TL0 is the most important to
       // improve since the errors in this layer will propagate to TL1.
       // Currently, reduce max qp by 20% for TL0 and 15% for TL1.
       layers_[0].enhanced_max_qp = min_qp_ + (((max_qp_ - min_qp_) * 80) / 100);
       layers_[1].enhanced_max_qp = min_qp_ + (((max_qp_ - min_qp_) * 85) / 100);
     }
 
-    if (incoming_framerate_) {
+    if (capture_framerate_) {
       int avg_frame_size =
-          (target_bitrate_kbps * 1000) / (8 * *incoming_framerate_);
+          (target_bitrate_kbps * 1000) / (8 * *capture_framerate_);
       max_debt_bytes_ = 4 * avg_frame_size;
     }
 
     bitrate_updated_ = false;
     cfg_updated = true;
   }
 
   // Don't try to update boosts state if not active yet.
   if (active_layer_ == -1)
     return cfg_updated;
@@ skipping 65 matching lines @@
       RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.Screenshare.Layer1.Qp",
                                  stats_.tl1_qp_sum_ / stats_.num_tl1_frames_);
       RTC_HISTOGRAM_COUNTS_10000(
           "WebRTC.Video.Screenshare.Layer1.TargetBitrate",
           stats_.tl1_target_bitrate_sum_ / stats_.num_tl1_frames_);
     }
   }
 }
 
 }  // namespace webrtc