Index: webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
index a806b3db0e663149c1157a7833fb306304210847..c79fe402e0b9c0f9eb6faa3e643ea382ac187cfa 100644 |
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc |
@@ -22,6 +22,7 @@ |
#include "webrtc/base/logging.h" |
#include "webrtc/call/call.h" |
#include "webrtc/logging/rtc_event_log/rtc_event_log.h" |
+#include "webrtc/modules/audio_coding/audio_network_adaptor/include/audio_network_adaptor.h" |
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h" |
namespace webrtc { |
@@ -511,7 +512,7 @@ void ParsedRtcEventLog::GetDelayBasedBweUpdate( |
void ParsedRtcEventLog::GetAudioNetworkAdaptation( |
size_t index, |
- AudioNetworkAdaptor::EncoderRuntimeConfig* config) const { |
+ AudioEncoderRuntimeConfig* config) const { |
RTC_CHECK_LT(index, GetNumberOfEvents()); |
const rtclog::Event& event = events_[index]; |
RTC_CHECK(event.has_type()); |