Loosening the coupling between WebRTC and //third_party/protobuf

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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
 #include <algorithm>
 #include <iterator>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/numerics/exp_filter.h"
+#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/base/safe_conversions.h"
 #include "webrtc/base/timeutils.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/audio_network_adaptor_impl.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 #include "webrtc/system_wrappers/include/field_trial.h"
 
 namespace webrtc {
 
@@ skipping 156 matching lines @@
     std::unique_ptr<SmoothingFilter> bitrate_smoother)
     : send_side_bwe_with_overhead_(webrtc::field_trial::IsEnabled(
           "WebRTC-SendSideBwe-WithOverhead")),
       packet_loss_rate_(0.0),
       inst_(nullptr),
       packet_loss_fraction_smoother_(new PacketLossFractionSmoother(
           config.clock)),
       audio_network_adaptor_creator_(
           audio_network_adaptor_creator
               ? std::move(audio_network_adaptor_creator)
-              : [this](const std::string& config_string,
+              : [this](const ProtoString& config_string,
                        RtcEventLog* event_log,
                        const Clock* clock) {
                   return DefaultAudioNetworkAdaptorCreator(config_string,
                                                            event_log, clock);
               }),
       bitrate_smoother_(bitrate_smoother
           ? std::move(bitrate_smoother) : std::unique_ptr<SmoothingFilter>(
               // We choose 5sec as initial time constant due to empirical data.
               new SmoothingFilterImpl(5000, config.clock))) {
   RTC_CHECK(RecreateEncoderInstance(config));
@@ skipping 335 matching lines @@
   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 ProtoString& 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(),
           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