Make a the decisions of ANA optional for the opus encoder.

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 473 matching lines @@
   RTC_CHECK_EQ(0, WebRtcOpus_SetBitRate(inst_, config_.GetBitrateBps()));
   const auto new_complexity = config_.GetNewComplexity();
   if (new_complexity && complexity_ != *new_complexity) {
     complexity_ = *new_complexity;
     RTC_CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, complexity_));
   }
 }
 
 void AudioEncoderOpus::ApplyAudioNetworkAdaptor() {
   auto config = audio_network_adaptor_->GetEncoderRuntimeConfig();
-  // |audio_network_adaptor_| is supposed to be configured to output all
-  // following parameters.
-  RTC_DCHECK(config.bitrate_bps);
-  RTC_DCHECK(config.frame_length_ms);
-  RTC_DCHECK(config.uplink_packet_loss_fraction);
-  RTC_DCHECK(config.enable_fec);
-  RTC_DCHECK(config.enable_dtx);
-  RTC_DCHECK(config.num_channels);
+  RTC_DCHECK(!config.frame_length_ms || *config.frame_length_ms == 20 ||
+             *config.frame_length_ms == 60);
 
-  RTC_DCHECK(*config.frame_length_ms == 20 || *config.frame_length_ms == 60);
-
-  SetTargetBitrate(*config.bitrate_bps);
-  SetFrameLength(*config.frame_length_ms);
-  SetFec(*config.enable_fec);
-  SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
-  SetDtx(*config.enable_dtx);
-  SetNumChannelsToEncode(*config.num_channels);
+  if (config.bitrate_bps)
+    SetTargetBitrate(*config.bitrate_bps);
+  if (config.frame_length_ms)
+    SetFrameLength(*config.frame_length_ms);
+  if (config.enable_fec)
+    SetFec(*config.enable_fec);
+  if (config.uplink_packet_loss_fraction)
+    SetProjectedPacketLossRate(*config.uplink_packet_loss_fraction);
+  if (config.enable_dtx)
+    SetDtx(*config.enable_dtx);
+  if (config.num_channels)
+    SetNumChannelsToEncode(*config.num_channels);
 }
 
 std::unique_ptr<AudioNetworkAdaptor>
 AudioEncoderOpus::DefaultAudioNetworkAdaptorCreator(
     const std::string& config_string,
     const Clock* clock) const {
   AudioNetworkAdaptorImpl::Config config;
   config.clock = clock;
   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)));
 }
 
 }  // namespace webrtc