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

webrtc/modules/audio_processing/audio_processing_impl.cc

 /*
  *  Copyright (c) 2012 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 1860 matching lines @@
   debug_state->event_msg->Clear();
 
   return kNoError;
 }
 
 int AudioProcessingImpl::WriteInitMessage() {
   debug_dump_.capture.event_msg->set_type(audioproc::Event::INIT);
   audioproc::Init* msg = debug_dump_.capture.event_msg->mutable_init();
   msg->set_sample_rate(formats_.api_format.input_stream().sample_rate_hz());
 
-  msg->set_num_input_channels(static_cast<google::protobuf::int32>(
+  msg->set_num_input_channels(static_cast<int32_t>(
       formats_.api_format.input_stream().num_channels()));
-  msg->set_num_output_channels(static_cast<google::protobuf::int32>(
+  msg->set_num_output_channels(static_cast<int32_t>(
       formats_.api_format.output_stream().num_channels()));
-  msg->set_num_reverse_channels(static_cast<google::protobuf::int32>(
+  msg->set_num_reverse_channels(static_cast<int32_t>(
       formats_.api_format.reverse_input_stream().num_channels()));
   msg->set_reverse_sample_rate(
       formats_.api_format.reverse_input_stream().sample_rate_hz());
   msg->set_output_sample_rate(
       formats_.api_format.output_stream().sample_rate_hz());
   msg->set_reverse_output_sample_rate(
       formats_.api_format.reverse_output_stream().sample_rate_hz());
   msg->set_num_reverse_output_channels(
       formats_.api_format.reverse_output_stream().num_channels());
 
@@ skipping 49 matching lines @@
     experiments_description += "LevelController;";
   }
   if (constants_.agc_clipped_level_min != kClippedLevelMin) {
     experiments_description += "AgcClippingLevelExperiment;";
   }
   if (capture_nonlocked_.echo_canceller3_enabled) {
     experiments_description += "EchoCanceller3;";
   }
   config.set_experiments_description(experiments_description);
 
-  std::string serialized_config = config.SerializeAsString();
+  ProtoString serialized_config = config.SerializeAsString();
   if (!forced &&
       debug_dump_.capture.last_serialized_config == serialized_config) {
     return kNoError;
   }
 
   debug_dump_.capture.last_serialized_config = serialized_config;
 
   debug_dump_.capture.event_msg->set_type(audioproc::Event::CONFIG);
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
@@ skipping 22 matching lines @@
       capture_processing_format(kSampleRate16kHz),
       split_rate(kSampleRate16kHz) {}
 
 AudioProcessingImpl::ApmCaptureState::~ApmCaptureState() = default;
 
 AudioProcessingImpl::ApmRenderState::ApmRenderState() = default;
 
 AudioProcessingImpl::ApmRenderState::~ApmRenderState() = default;
 
 }  // namespace webrtc