| Index: webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
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| diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
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| index feffff0807439b1bdfa5a1d2b886a3cb42516441..f8148b2951e7600b2ae08c186a0288bc879db3d3 100644
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| --- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
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| +++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
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| @@ -490,10 +490,8 @@ void ParsedRtcEventLog::GetLossBasedBweUpdate(size_t index,
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|    }
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|  }
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|  
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| -void ParsedRtcEventLog::GetDelayBasedBweUpdate(
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| -    size_t index,
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| -    int32_t* bitrate_bps,
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| -    BandwidthUsage* detector_state) const {
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| +ParsedRtcEventLog::BweDelayBasedUpdate
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| +ParsedRtcEventLog::GetDelayBasedBweUpdate(size_t index) const {
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|    RTC_CHECK_LT(index, GetNumberOfEvents());
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|    const rtclog::Event& event = events_[index];
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|    RTC_CHECK(event.has_type());
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| @@ -501,14 +499,14 @@ void ParsedRtcEventLog::GetDelayBasedBweUpdate(
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|    RTC_CHECK(event.has_delay_based_bwe_update());
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|    const rtclog::DelayBasedBweUpdate& delay_event =
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|        event.delay_based_bwe_update();
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| +
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| +  BweDelayBasedUpdate res;
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| +  res.timestamp = GetTimestamp(index);
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|    RTC_CHECK(delay_event.has_bitrate_bps());
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| -  if (bitrate_bps != nullptr) {
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| -    *bitrate_bps = delay_event.bitrate_bps();
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| -  }
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| +  res.bitrate_bps = delay_event.bitrate_bps();
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|    RTC_CHECK(delay_event.has_detector_state());
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| -  if (detector_state != nullptr) {
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| -    *detector_state = GetRuntimeDetectorState(delay_event.detector_state());
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| -  }
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| +  res.detector_state = GetRuntimeDetectorState(delay_event.detector_state());
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| +  return res;
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|  }
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|  
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|  void ParsedRtcEventLog::GetAudioNetworkAdaptation(
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| 
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