Experiment-driven configuration of PLR/RPLR-based FecController

webrtc/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.
  */
 
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 
 #include <cmath>
 #include <utility>
 
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_plr_based.h"
+#include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller_rplr_based.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
 RTC_PUSH_IGNORING_WUNDEF()
 #ifdef WEBRTC_ANDROID_PLATFORM_BUILD
 #include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #else
 #include "webrtc/modules/audio_coding/audio_network_adaptor/config.pb.h"
 #endif
@@ skipping 35 matching lines @@
               fec_enabling_threshold.high_bandwidth_bps(),
               fec_enabling_threshold.high_bandwidth_packet_loss()),
           FecControllerPlrBased::Config::Threshold(
               fec_disabling_threshold.low_bandwidth_bps(),
               fec_disabling_threshold.low_bandwidth_packet_loss(),
               fec_disabling_threshold.high_bandwidth_bps(),
               fec_disabling_threshold.high_bandwidth_packet_loss()),
           config.time_constant_ms(), clock)));
 }
 
+std::unique_ptr<FecControllerRplrBased> CreateFecControllerRplrBased(
+    const audio_network_adaptor::config::FecControllerRplrBased& config,
+    bool initial_fec_enabled,
+    const Clock* clock) {
+  RTC_CHECK(config.has_fec_enabling_threshold());
+  RTC_CHECK(config.has_fec_disabling_threshold());
+  RTC_CHECK(config.has_time_constant_ms());
+
+  auto& fec_enabling_threshold = config.fec_enabling_threshold();
+  RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_bps());
+  RTC_CHECK(fec_enabling_threshold.has_low_bandwidth_recoverable_packet_loss());
+  RTC_CHECK(fec_enabling_threshold.has_high_bandwidth_bps());
+  RTC_CHECK(
+      fec_enabling_threshold.has_high_bandwidth_recoverable_packet_loss());
+
+  auto& fec_disabling_threshold = config.fec_disabling_threshold();
+  RTC_CHECK(fec_disabling_threshold.has_low_bandwidth_bps());
+  RTC_CHECK(
+      fec_disabling_threshold.has_low_bandwidth_recoverable_packet_loss());
+  RTC_CHECK(fec_disabling_threshold.has_high_bandwidth_bps());
+  RTC_CHECK(
+      fec_disabling_threshold.has_high_bandwidth_recoverable_packet_loss());
+
+  return std::unique_ptr<FecControllerRplrBased>(
+      new FecControllerRplrBased(FecControllerRplrBased::Config(
+          initial_fec_enabled,
+          FecControllerRplrBased::Config::Threshold(
+              fec_enabling_threshold.low_bandwidth_bps(),
+              fec_enabling_threshold.low_bandwidth_recoverable_packet_loss(),
+              fec_enabling_threshold.high_bandwidth_bps(),
+              fec_enabling_threshold.high_bandwidth_recoverable_packet_loss()),
+          FecControllerRplrBased::Config::Threshold(
+              fec_disabling_threshold.low_bandwidth_bps(),
+              fec_disabling_threshold.low_bandwidth_recoverable_packet_loss(),
+              fec_disabling_threshold.high_bandwidth_bps(),
+              fec_disabling_threshold.high_bandwidth_recoverable_packet_loss()),
+          config.time_constant_ms(), clock)));
+}
+
 std::unique_ptr<FrameLengthController> CreateFrameLengthController(
     const audio_network_adaptor::config::FrameLengthController& config,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
     int initial_frame_length_ms,
     int min_encoder_bitrate_bps) {
   RTC_CHECK(config.has_fl_increasing_packet_loss_fraction());
   RTC_CHECK(config.has_fl_decreasing_packet_loss_fraction());
   RTC_CHECK(config.has_fl_20ms_to_60ms_bandwidth_bps());
   RTC_CHECK(config.has_fl_60ms_to_20ms_bandwidth_bps());
 
@@ skipping 90 matching lines @@
   std::map<const Controller*, std::pair<int, float>> chracteristic_points;
 
   for (int i = 0; i < controller_manager_config.controllers_size(); ++i) {
     auto& controller_config = controller_manager_config.controllers(i);
     std::unique_ptr<Controller> controller;
     switch (controller_config.controller_case()) {
       case audio_network_adaptor::config::Controller::kFecController:
         controller = CreateFecControllerPlrBased(
             controller_config.fec_controller(), initial_fec_enabled, clock);
         break;
+      case audio_network_adaptor::config::Controller::kFecControllerRplrBased:
+        controller = CreateFecControllerRplrBased(
+            controller_config.fec_controller_rplr_based(), initial_fec_enabled,
+            clock);
+        break;
       case audio_network_adaptor::config::Controller::kFrameLengthController:
         controller = CreateFrameLengthController(
             controller_config.frame_length_controller(),
             encoder_frame_lengths_ms, initial_frame_length_ms,
             min_encoder_bitrate_bps);
         break;
       case audio_network_adaptor::config::Controller::kChannelController:
         controller = CreateChannelController(
             controller_config.channel_controller(), num_encoder_channels,
             intial_channels_to_encode);
@@ skipping 149 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