Added configurable offsets to the per-packet overhead in ANA.

modules/audio_coding/audio_network_adaptor/controller_manager.cc

 /*
  *  Copyright (c) 2016 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 119 matching lines @@
   if (config.has_fl_60ms_to_120ms_bandwidth_bps() &&
       config.has_fl_120ms_to_60ms_bandwidth_bps()) {
     fl_changing_bandwidths_bps.insert(std::make_pair(
         FrameLengthController::Config::FrameLengthChange(60, 120),
         config.fl_60ms_to_120ms_bandwidth_bps()));
     fl_changing_bandwidths_bps.insert(std::make_pair(
         FrameLengthController::Config::FrameLengthChange(120, 60),
         config.fl_120ms_to_60ms_bandwidth_bps()));
   }
 
+  int fl_increase_overhead_offset = 0;
+  if (config.has_fl_increase_overhead_offset()) {
+    fl_increase_overhead_offset = config.fl_increase_overhead_offset();
+  }
+  int fl_decrease_overhead_offset = 0;
+  if (config.has_fl_decrease_overhead_offset()) {
+    fl_decrease_overhead_offset = config.fl_decrease_overhead_offset();
+  }
+
   FrameLengthController::Config ctor_config(
       std::vector<int>(), initial_frame_length_ms, min_encoder_bitrate_bps,
       config.fl_increasing_packet_loss_fraction(),
-      config.fl_decreasing_packet_loss_fraction(),
-      std::move(fl_changing_bandwidths_bps));
+      config.fl_decreasing_packet_loss_fraction(), fl_increase_overhead_offset,
+      fl_decrease_overhead_offset, std::move(fl_changing_bandwidths_bps));
 
   for (auto frame_length : encoder_frame_lengths_ms)
     ctor_config.encoder_frame_lengths_ms.push_back(frame_length);
 
   return std::unique_ptr<FrameLengthController>(
       new FrameLengthController(ctor_config));
 }
 
 std::unique_ptr<ChannelController> CreateChannelController(
     const audio_network_adaptor::config::ChannelController& config,
@@ skipping 14 matching lines @@
   RTC_CHECK(dtx_config.has_dtx_enabling_bandwidth_bps());
   RTC_CHECK(dtx_config.has_dtx_disabling_bandwidth_bps());
 
   return std::unique_ptr<DtxController>(new DtxController(DtxController::Config(
       initial_dtx_enabled, dtx_config.dtx_enabling_bandwidth_bps(),
       dtx_config.dtx_disabling_bandwidth_bps())));
 }
 
 using audio_network_adaptor::BitrateController;
 std::unique_ptr<BitrateController> CreateBitrateController(
+    const audio_network_adaptor::config::BitrateController& bitrate_config,
     int initial_bitrate_bps,
     int initial_frame_length_ms) {
-  return std::unique_ptr<BitrateController>(new BitrateController(
-      BitrateController::Config(initial_bitrate_bps, initial_frame_length_ms)));
+  int fl_increase_overhead_offset = 0;
+  if (bitrate_config.has_fl_increase_overhead_offset()) {
+    fl_increase_overhead_offset = bitrate_config.fl_increase_overhead_offset();
+  }
+  int fl_decrease_overhead_offset = 0;
+  if (bitrate_config.has_fl_decrease_overhead_offset()) {
+    fl_decrease_overhead_offset = bitrate_config.fl_decrease_overhead_offset();
+  }
+  return std::unique_ptr<BitrateController>(
+      new BitrateController(BitrateController::Config(
+          initial_bitrate_bps, initial_frame_length_ms,
+          fl_increase_overhead_offset, fl_decrease_overhead_offset)));
 }
 #endif  // WEBRTC_ENABLE_PROTOBUF
 
 }  // namespace
 
 ControllerManagerImpl::Config::Config(int min_reordering_time_ms,
                                       float min_reordering_squared_distance)
     : min_reordering_time_ms(min_reordering_time_ms),
       min_reordering_squared_distance(min_reordering_squared_distance) {}
 
@@ skipping 57 matching lines @@
       case audio_network_adaptor::config::Controller::kChannelController:
         controller = CreateChannelController(
             controller_config.channel_controller(), num_encoder_channels,
             intial_channels_to_encode);
         break;
       case audio_network_adaptor::config::Controller::kDtxController:
         controller = CreateDtxController(controller_config.dtx_controller(),
                                          initial_dtx_enabled);
         break;
       case audio_network_adaptor::config::Controller::kBitrateController:
-        controller = CreateBitrateController(initial_bitrate_bps,
-                                             initial_frame_length_ms);
+        controller = CreateBitrateController(
+            controller_config.bitrate_controller(), initial_bitrate_bps,
+            initial_frame_length_ms);
         break;
       default:
         RTC_NOTREACHED();
     }
     if (controller_config.has_scoring_point()) {
       auto& scoring_point = controller_config.scoring_point();
       RTC_CHECK(scoring_point.has_uplink_bandwidth_bps());
       RTC_CHECK(scoring_point.has_uplink_packet_loss_fraction());
       scoring_points[controller.get()] = std::make_pair<int, float>(
           scoring_point.uplink_bandwidth_bps(),
@@ skipping 139 matching lines @@
       NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
       NormalizeUplinkBandwidth(uplink_bandwidth_bps);
   float diff_normalized_packet_loss =
       NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
       NormalizePacketLossFraction(uplink_packet_loss_fraction);
   return std::pow(diff_normalized_bitrate_bps, 2) +
          std::pow(diff_normalized_packet_loss, 2);
 }
 
 }  // namespace webrtc