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..15effe87b911456aed6558a20b94f2810749620b 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 |
@@ -14,6 +14,7 @@ |
#include "webrtc/base/ignore_wundef.h" |
#ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP |
+#include "webrtc/base/protobuf_utils.h" |
RTC_PUSH_IGNORING_WUNDEF() |
#ifdef WEBRTC_ANDROID_PLATFORM_BUILD |
#include "external/webrtc/webrtc/modules/audio_coding/audio_network_adaptor/debug_dump.pb.h" |
@@ -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)); |