Add 120ms frame ability to ANA

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 62 matching lines @@
 }
 
 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) {
   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());
+  RTC_CHECK(config.has_fl_60ms_to_120ms_bandwidth_bps());

[minyue-webrtc, 2017/02/02 08:31:10]

this must be carefully treated for backward-compatibility.

[michaelt, 2017/02/02 10:29:57]

Impl. as discussed offline. 

+  RTC_CHECK(config.has_fl_120ms_to_60ms_bandwidth_bps());
 
   FrameLengthController::Config ctor_config(
       std::vector<int>(), initial_frame_length_ms,
       config.fl_increasing_packet_loss_fraction(),
       config.fl_decreasing_packet_loss_fraction(),
       config.fl_20ms_to_60ms_bandwidth_bps(),
-      config.fl_60ms_to_20ms_bandwidth_bps());
+      config.fl_60ms_to_20ms_bandwidth_bps(),
+      config.fl_60ms_to_120ms_bandwidth_bps(),
+      config.fl_120ms_to_60ms_bandwidth_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 225 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