| Index: webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
|
| 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
|
| index 7770e65a269ec8e246d982e647f9ad3bc2d23f23..32724bac8a6d41660349b602deddd08a33235209 100644
|
| --- a/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
|
| +++ b/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump_writer.cc
|
| @@ -12,6 +12,7 @@
|
|
|
| #include "webrtc/base/checks.h"
|
| #include "webrtc/base/ignore_wundef.h"
|
| +#include "webrtc/base/protobuf_utils.h"
|
|
|
| #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
|
| RTC_PUSH_IGNORING_WUNDEF()
|
| @@ -34,7 +35,7 @@ using audio_network_adaptor::debug_dump::EncoderRuntimeConfig;
|
|
|
| void DumpEventToFile(const Event& event, FileWrapper* dump_file) {
|
| RTC_CHECK(dump_file->is_open());
|
| - std::string dump_data;
|
| + ProtoString dump_data;
|
| event.SerializeToString(&dump_data);
|
| int32_t size = event.ByteSize();
|
| dump_file->Write(&size, sizeof(size));
|
|
|