Change frame length on very low bandwidth.

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.
  */
 
@@ skipping 57 matching lines @@
           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<FrameLengthController> CreateFrameLengthController(
     const audio_network_adaptor::config::FrameLengthController& config,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
-    int initial_frame_length_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());
 
   std::map<FrameLengthController::Config::FrameLengthChange, int>
       fl_changing_bandwidths_bps = {
           {FrameLengthController::Config::FrameLengthChange(20, 60),
            config.fl_20ms_to_60ms_bandwidth_bps()},
           {FrameLengthController::Config::FrameLengthChange(60, 20),
            config.fl_60ms_to_20ms_bandwidth_bps()}};
 
   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()));
   }
 
+  constexpr int kPreventOveruseMarginBps = 5000;
   FrameLengthController::Config ctor_config(
-      std::vector<int>(), initial_frame_length_ms,
-      config.fl_increasing_packet_loss_fraction(),
+      std::vector<int>(), initial_frame_length_ms, min_encoder_bitrate_bps,
+      kPreventOveruseMarginBps, config.fl_increasing_packet_loss_fraction(),
       config.fl_decreasing_packet_loss_fraction(),
       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));
 }
 
@@ skipping 38 matching lines @@
     : min_reordering_time_ms(min_reordering_time_ms),
       min_reordering_squared_distance(min_reordering_squared_distance),
       clock(clock) {}
 
 ControllerManagerImpl::Config::~Config() = default;
 
 std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
     const std::string& config_string,
     size_t num_encoder_channels,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
+    int min_encoder_bitrate_bps,
     size_t intial_channels_to_encode,
     int initial_frame_length_ms,
     int initial_bitrate_bps,
     bool initial_fec_enabled,
     bool initial_dtx_enabled,
     const Clock* clock) {
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
   audio_network_adaptor::config::ControllerManager controller_manager_config;
   controller_manager_config.ParseFromString(config_string);
 
   std::vector<std::unique_ptr<Controller>> controllers;
   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 = CreateFecController(controller_config.fec_controller(),
                                          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);
+            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);
         break;
       case audio_network_adaptor::config::Controller::kDtxController:
         controller = CreateDtxController(controller_config.dtx_controller(),
                                          initial_dtx_enabled);
         break;
@@ skipping 144 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