| 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());
|
|
|