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 2e4757ab8ccd1b4928cc79f0624999746cf3ca6e..8b485e4a075f11bea63c1cfd47ea64f166acd2b3 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)); |