| 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..6b948e0152018082e1f52755b2e8d2850fa0a871 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
|
| @@ -23,6 +23,8 @@ RTC_PUSH_IGNORING_WUNDEF()
|
| RTC_POP_IGNORING_WUNDEF()
|
| #endif
|
|
|
| +#include "webrtc/base/protobuf_utils.h"
|
| +
|
| namespace webrtc {
|
|
|
| #ifdef WEBRTC_AUDIO_NETWORK_ADAPTOR_DEBUG_DUMP
|
| @@ -34,7 +36,7 @@ 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));
|
|
|