| Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
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| diff --git a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
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| index 7770e65a269ec8e246d982e647f9ad3bc2d23f23..6b948e0152018082e1f52755b2e8d2850fa0a871 100644
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| --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
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| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
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| @@ -23,6 +23,8 @@ RTC_PUSH_IGNORING_WUNDEF()
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|  RTC_POP_IGNORING_WUNDEF()
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|  #endif
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|  
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| +#include "webrtc/base/protobuf_utils.h"
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| +
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|  namespace webrtc {
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|  
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|  #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
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| @@ -34,7 +36,7 @@ using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
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|  
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|  void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
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|    RTC_CHECK(dump_file->is_open());
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| -  std::string dump_data;
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| +  ProtoString dump_data;
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|    event.SerializeToString(&dump_data);
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|    int32_t size = event.ByteSize();
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|    dump_file->Write(&size, sizeof(size));
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| 
 |