Loosening the coupling between WebRTC and //third_party/protobuf

webrtc/modules/audio_coding/audio_network_adaptor/controller_manager.cc

 /*
  *  Copyright (c) 2016 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/audio_network_adaptor/controller_manager.h"
 
 #include <cmath>
 #include <utility>
+#include <string>

[kjellander_webrtc, 2017/03/15 12:02:30]

Sort alphabetically.

[mbonadei, 2017/03/15 13:21:48]

Done.

 
 #include "webrtc/base/ignore_wundef.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/bitrate_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/channel_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/dtx_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/fec_controller.h"
 #include "webrtc/modules/audio_coding/audio_network_adaptor/frame_length_controller.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
@@ skipping 125 matching lines @@
 ControllerManagerImpl::Config::Config(int min_reordering_time_ms,
                                       float min_reordering_squared_distance,
                                       const Clock* clock)
     : min_reordering_time_ms(min_reordering_time_ms),
       min_reordering_squared_distance(min_reordering_squared_distance),
       clock(clock) {}
 
 ControllerManagerImpl::Config::~Config() = default;
 
 std::unique_ptr<ControllerManager> ControllerManagerImpl::Create(
+#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
+    const ProtoString& config_string,
+#else
     const std::string& config_string,
+#endif
     size_t num_encoder_channels,
     rtc::ArrayView<const int> encoder_frame_lengths_ms,
     int min_encoder_bitrate_bps,
     size_t intial_channels_to_encode,
     int initial_frame_length_ms,
     int initial_bitrate_bps,
     bool initial_fec_enabled,
     bool initial_dtx_enabled,
     const Clock* clock) {
 #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
@@ skipping 173 matching lines @@
       NormalizeUplinkBandwidth(scoring_point.uplink_bandwidth_bps) -
       NormalizeUplinkBandwidth(uplink_bandwidth_bps);
   float diff_normalized_packet_loss =
       NormalizePacketLossFraction(scoring_point.uplink_packet_loss_fraction) -
       NormalizePacketLossFraction(uplink_packet_loss_fraction);
   return std::pow(diff_normalized_bitrate_bps, 2) +
          std::pow(diff_normalized_packet_loss, 2);
 }
 
 }  // namespace webrtc