| 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 feffff0807439b1bdfa5a1d2b886a3cb42516441..a6ff8af80079786028d0064640ec9d2432d2e056 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
|
| @@ -24,6 +24,7 @@
|
| #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/remote_bitrate_estimator/include/bwe_defines.h"
|
| #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
|
|
|
| namespace webrtc {
|
| @@ -97,14 +98,14 @@ BandwidthUsage GetRuntimeDetectorState(
|
| rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
|
| switch (detector_state) {
|
| case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
|
| - return kBwNormal;
|
| + return BandwidthUsage::kBwNormal;
|
| case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
|
| - return kBwUnderusing;
|
| + return BandwidthUsage::kBwUnderusing;
|
| case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
|
| - return kBwOverusing;
|
| + return BandwidthUsage::kBwOverusing;
|
| }
|
| RTC_NOTREACHED();
|
| - return kBwNormal;
|
| + return BandwidthUsage::kBwNormal;
|
| }
|
|
|
| std::pair<uint64_t, bool> ParseVarInt(std::istream& stream) {
|
|
|