Add frame rate throttling to vp8 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 58 matching lines @@
       number_of_temporal_layers_(num_temporal_layers),
       last_base_layer_sync_(false),
       tl0_pic_idx_(initial_tl0_pic_idx),
       active_layer_(-1),
       last_timestamp_(-1),
       last_sync_timestamp_(-1),
       last_emitted_tl0_timestamp_(-1),
       min_qp_(-1),
       max_qp_(-1),
       max_debt_bytes_(0),
-      framerate_(-1),
+      encode_framerate_(1000.0f, 1000.0f),  // 1 second window, second scale.
       bitrate_updated_(false) {
   RTC_CHECK_GT(num_temporal_layers, 0);
   RTC_CHECK_LE(num_temporal_layers, 2);
 }
 
 ScreenshareLayers::~ScreenshareLayers() {
   UpdateHistograms();
 }
 
 int ScreenshareLayers::CurrentLayerId() const {
   // Codec does not use temporal layers for screenshare.
   return 0;
 }
 
 int ScreenshareLayers::EncodeFlags(uint32_t timestamp) {
   if (number_of_temporal_layers_ <= 1) {
     // No flags needed for 1 layer screenshare.
     return 0;
   }
 
+  const int64_t now_ms = clock_->TimeInMilliseconds();
+  if (target_framerate_.value_or(0) > 0 &&
+      encode_framerate_.Rate(now_ms).value_or(0) > *target_framerate_) {
+    // Max framerate exceeded, drop frame.
+    return -1;
+  }
+
   if (stats_.first_frame_time_ms_ == -1)
-    stats_.first_frame_time_ms_ = clock_->TimeInMilliseconds();
+    stats_.first_frame_time_ms_ = now_ms;
 
   int64_t unwrapped_timestamp = time_wrap_handler_.Unwrap(timestamp);
   int flags = 0;
   if (active_layer_ == -1 ||
       layers_[active_layer_].state != TemporalLayer::State::kDropped) {
     if (last_emitted_tl0_timestamp_ != -1 &&
         (unwrapped_timestamp - last_emitted_tl0_timestamp_) / 90 >
             kMaxFrameIntervalMs) {
       // Too long time has passed since the last frame was emitted, cancel
       // enough debt to allow a single frame.
@@ skipping 29 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 / (framerate_ <= 0 ? 5 : framerate_);
+    ts_diff =
+        kOneSecond90Khz / incoming_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) {
-  bitrate_updated_ =
-      bitrate_kbps != static_cast<int>(layers_[0].target_rate_kbps_) ||
-      max_bitrate_kbps != static_cast<int>(layers_[1].target_rate_kbps_) ||
-      framerate != 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_;
+    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_));
+    if (framerate < 0) {
+      incoming_framerate_.reset();
+    } else {
+      incoming_framerate_.emplace(framerate);
+    }
+  }
+
   layers_[0].target_rate_kbps_ = bitrate_kbps;
   layers_[1].target_rate_kbps_ = max_bitrate_kbps;
-  framerate_ = framerate;
 
   std::vector<uint32_t> allocation;
   allocation.push_back(bitrate_kbps);
   if (max_bitrate_kbps > bitrate_kbps)
     allocation.push_back(max_bitrate_kbps - bitrate_kbps);
   return allocation;
 }
 
 void ScreenshareLayers::FrameEncoded(unsigned int size,
                                      uint32_t timestamp,
                                      int qp) {
+  if (size > 0)
+    encode_framerate_.Update(1, clock_->TimeInMilliseconds());
+
   if (number_of_temporal_layers_ == 1)
     return;
 
   RTC_DCHECK_NE(-1, active_layer_);
   if (size == 0) {
     layers_[active_layer_].state = TemporalLayer::State::kDropped;
     ++stats_.num_overshoots_;
     return;
   }
 
@@ skipping 101 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 (framerate_ > 0) {
-      int avg_frame_size = (target_bitrate_kbps * 1000) / (8 * framerate_);
+    if (incoming_framerate_) {

[hta-webrtc, 2017/01/11 08:19:46]

Using the term "incoming" here is not a Good Thing.
It's usually used to refer to "stuff coming from the other end of the
connection".

Please consider renaming to "source_framerate_".

[sprang_webrtc, 2017/01/11 11:41:59]

I create a new cl for that: https://codereview.webrtc.org/2624073003/

+      int avg_frame_size =
+          (target_bitrate_kbps * 1000) / (8 * *incoming_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