Reland of Loosening the coupling between WebRTC and //third_party/protobuf

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

 /*
  *  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.
  */
 
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_
 
 #include <functional>
 #include <memory>
 #include <string>
 #include <vector>
 
 #include "webrtc/api/audio_codecs/audio_format.h"
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/optional.h"
+#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/common_audio/smoothing_filter.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/opus/opus_interface.h"
 
 namespace webrtc {
 
 class RtcEventLog;
 
 struct CodecInst;
@@ skipping 128 matching lines @@
   void SetFrameLength(int frame_length_ms);
   void SetNumChannelsToEncode(size_t num_channels_to_encode);
   void SetProjectedPacketLossRate(float fraction);
 
   // TODO(minyue): remove "override" when we can deprecate
   // |AudioEncoder::SetTargetBitrate|.
   void SetTargetBitrate(int target_bps) override;
 
   void ApplyAudioNetworkAdaptor();
   std::unique_ptr<AudioNetworkAdaptor> DefaultAudioNetworkAdaptorCreator(
-      const std::string& config_string,
+      const ProtoString& config_string,
       RtcEventLog* event_log,
       const Clock* clock) const;
 
   void MaybeUpdateUplinkBandwidth();
 
   Config config_;
   const bool send_side_bwe_with_overhead_;
   float packet_loss_rate_;
   std::vector<int16_t> input_buffer_;
   OpusEncInst* inst_;
   uint32_t first_timestamp_in_buffer_;
   size_t num_channels_to_encode_;
   int next_frame_length_ms_;
   int complexity_;
   std::unique_ptr<PacketLossFractionSmoother> packet_loss_fraction_smoother_;
   AudioNetworkAdaptorCreator audio_network_adaptor_creator_;
   std::unique_ptr<AudioNetworkAdaptor> audio_network_adaptor_;
   rtc::Optional<size_t> overhead_bytes_per_packet_;
   const std::unique_ptr<SmoothingFilter> bitrate_smoother_;
   rtc::Optional<int64_t> bitrate_smoother_last_update_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderOpus);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_OPUS_AUDIO_ENCODER_OPUS_H_