Change frame length on very low bandwidth.

webrtc/modules/audio_coding/audio_network_adaptor/config.proto

 syntax = "proto2";
 option optimize_for = LITE_RUNTIME;
 option java_package = "org.webrtc.AudioNetworkAdaptor";
 option java_outer_classname = "Config";
 package webrtc.audio_network_adaptor.config;
 
 message FecController {
   message Threshold {
     // Threshold defines a curve in the bandwidth/packet-loss domain. The
     // curve is characterized by the two conjunction points: A and B.
@@ skipping 25 matching lines @@
 
   // |time_constant_ms| is the time constant for an exponential filter, which
   // is used for smoothing the packet loss fraction.
   optional int32 time_constant_ms = 3;
 }
 
 message FrameLengthController {
   // Uplink packet loss fraction below which frame length can increase.
   optional float fl_increasing_packet_loss_fraction = 1;
 
-  // Uplink packet loss fraction below which frame length should decrease.
+  // Uplink packet loss fraction above which frame length should decrease.
   optional float fl_decreasing_packet_loss_fraction = 2;
 
   // Uplink bandwidth below which frame length can switch from 20ms to 60ms.
   optional int32 fl_20ms_to_60ms_bandwidth_bps = 3;
 
   // Uplink bandwidth above which frame length should switch from 60ms to 20ms.
   optional int32 fl_60ms_to_20ms_bandwidth_bps = 4;
 
   // Uplink bandwidth below which frame length can switch from 60ms to 120ms.
   optional int32 fl_60ms_to_120ms_bandwidth_bps = 5;
@@ skipping 50 matching lines @@
 message ControllerManager {
   repeated Controller controllers = 1;
 
   // Least time since last reordering for a new reordering to be made.
   optional int32 min_reordering_time_ms = 2;
 
   // Least squared distance from last scoring point for a new reordering to be
   // made.
   optional float min_reordering_squared_distance = 3;
 }