Change frame length on very low bandwidth.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  Copyright (c) 2014 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 528 matching lines @@
 
 std::unique_ptr<AudioNetworkAdaptor>
 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
     const std::string& config_string,
     RtcEventLog* event_log,
     const Clock* clock) const {
   AudioNetworkAdaptorImpl::Config config;
   config.clock = clock;
   config.event_log = event_log;
   return std::unique_ptr<AudioNetworkAdaptor>(new AudioNetworkAdaptorImpl(
-      config, ControllerManagerImpl::Create(
-                  config_string, NumChannels(), supported_frame_lengths_ms(),
-                  num_channels_to_encode_, next_frame_length_ms_,
-                  GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock)));
+      config,
+      ControllerManagerImpl::Create(
+          config_string, NumChannels(), supported_frame_lengths_ms(),
+          kMinBitrateBps, num_channels_to_encode_, next_frame_length_ms_,
+          GetTargetBitrate(), config_.fec_enabled, GetDtx(), clock)));
 }
 
 void AudioEncoderOpus::MaybeUpdateUplinkBandwidth() {
   if (audio_network_adaptor_) {
     int64_t now_ms = rtc::TimeMillis();
     if (!bitrate_smoother_last_update_time_ ||
         now_ms - *bitrate_smoother_last_update_time_ >=
             config_.uplink_bandwidth_update_interval_ms) {
       rtc::Optional<float> smoothed_bitrate = bitrate_smoother_->GetAverage();
       if (smoothed_bitrate)
         audio_network_adaptor_->SetUplinkBandwidth(*smoothed_bitrate);
       bitrate_smoother_last_update_time_ = rtc::Optional<int64_t>(now_ms);
     }
   }
 }
 
 }  // namespace webrtc