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( |