Loosening the coupling between WebRTC and //third_party/protobuf

webrtc/logging/rtc_event_log/rtc_event_log_parser.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/logging/rtc_event_log/rtc_event_log_parser.h"
 
 #include <stdint.h>
 #include <string.h>
 
 #include <algorithm>
 #include <fstream>
 #include <istream>
 #include <utility>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
+#include "webrtc/base/protobuf_utils.h"
 #include "webrtc/call/call.h"
 #include "webrtc/logging/rtc_event_log/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 
 namespace webrtc {
 
 namespace {
 MediaType GetRuntimeMediaType(rtclog::MediaType media_type) {
   switch (media_type) {
     case rtclog::MediaType::ANY:
@@ skipping 87 matching lines @@
     varint |= static_cast<uint64_t>(byte & 0x7F) << (7 * bytes_read);
     if ((byte & 0x80) == 0) {
       return std::make_pair(varint, true);
     }
   }
   return std::make_pair(varint, false);
 }
 
 void GetHeaderExtensions(
     std::vector<RtpExtension>* header_extensions,
-    const google::protobuf::RepeatedPtrField<rtclog::RtpHeaderExtension>&
-        proto_header_extensions) {
+    const RepeatedPtrField<rtclog::RtpHeaderExtension>&
+    proto_header_extensions) {
   header_extensions->clear();
   for (auto& p : proto_header_extensions) {
     RTC_CHECK(p.has_name());
     RTC_CHECK(p.has_id());
     const std::string& name = p.name();
     int id = p.id();
     header_extensions->push_back(RtpExtension(name, id));
   }
 }
 
@@ skipping 386 matching lines @@
   if (ana_event.has_frame_length_ms())
     config->frame_length_ms = rtc::Optional<int>(ana_event.frame_length_ms());
   if (ana_event.has_num_channels())
     config->num_channels = rtc::Optional<size_t>(ana_event.num_channels());
   if (ana_event.has_uplink_packet_loss_fraction())
     config->uplink_packet_loss_fraction =
         rtc::Optional<float>(ana_event.uplink_packet_loss_fraction());
 }
 
 }  // namespace webrtc