| 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..f8148b2951e7600b2ae08c186a0288bc879db3d3 100644
|
| --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
|
| +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
|
| @@ -490,10 +490,8 @@ void ParsedRtcEventLog::GetLossBasedBweUpdate(size_t index,
|
| }
|
| }
|
|
|
| -void ParsedRtcEventLog::GetDelayBasedBweUpdate(
|
| - size_t index,
|
| - int32_t* bitrate_bps,
|
| - BandwidthUsage* detector_state) const {
|
| +ParsedRtcEventLog::BweDelayBasedUpdate
|
| +ParsedRtcEventLog::GetDelayBasedBweUpdate(size_t index) const {
|
| RTC_CHECK_LT(index, GetNumberOfEvents());
|
| const rtclog::Event& event = events_[index];
|
| RTC_CHECK(event.has_type());
|
| @@ -501,14 +499,14 @@ void ParsedRtcEventLog::GetDelayBasedBweUpdate(
|
| RTC_CHECK(event.has_delay_based_bwe_update());
|
| const rtclog::DelayBasedBweUpdate& delay_event =
|
| event.delay_based_bwe_update();
|
| +
|
| + BweDelayBasedUpdate res;
|
| + res.timestamp = GetTimestamp(index);
|
| RTC_CHECK(delay_event.has_bitrate_bps());
|
| - if (bitrate_bps != nullptr) {
|
| - *bitrate_bps = delay_event.bitrate_bps();
|
| - }
|
| + res.bitrate_bps = delay_event.bitrate_bps();
|
| RTC_CHECK(delay_event.has_detector_state());
|
| - if (detector_state != nullptr) {
|
| - *detector_state = GetRuntimeDetectorState(delay_event.detector_state());
|
| - }
|
| + res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
|
| + return res;
|
| }
|
|
|
| void ParsedRtcEventLog::GetAudioNetworkAdaptation(
|
|
|