Video/Screenshare loopback tool.

webrtc/logging/rtc_event_log/rtc_event_log_parser.cc

 /*
  *  Copyright (c) 2016 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 56 matching lines @@
     case rtclog::Event::AUDIO_RECEIVER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_RECEIVER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_SENDER_CONFIG_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_SENDER_CONFIG_EVENT;
     case rtclog::Event::AUDIO_NETWORK_ADAPTATION_EVENT:
       return ParsedRtcEventLog::EventType::AUDIO_NETWORK_ADAPTATION_EVENT;
     case rtclog::Event::BWE_PROBE_CLUSTER_CREATED_EVENT:
       return ParsedRtcEventLog::EventType::BWE_PROBE_CLUSTER_CREATED_EVENT;
     case rtclog::Event::BWE_PROBE_RESULT_EVENT:
       return ParsedRtcEventLog::EventType::BWE_PROBE_RESULT_EVENT;
+    case rtclog::Event::BWE_ACKED_BITRATE_EVENT:
+      return ParsedRtcEventLog::EventType::BWE_ACKED_BITRATE_EVENT;
+    case rtclog::Event::ALR_STATE_EVENT:
+      return ParsedRtcEventLog::EventType::ALR_STATE_EVENT;
+    case rtclog::Event::PACKET_QUEUE_TIME:
+      return ParsedRtcEventLog::EventType::PACKET_QUEUE_TIME;
   }
   RTC_NOTREACHED();
   return ParsedRtcEventLog::EventType::UNKNOWN_EVENT;
 }
 
 BandwidthUsage GetRuntimeDetectorState(
     rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
   switch (detector_state) {
     case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
       return BandwidthUsage::kBwNormal;
@@ skipping 549 matching lines @@
         rtc::Optional<ProbeFailureReason>(kInvalidSendReceiveRatio);
   } else if (pr_event.result() == rtclog::BweProbeResult::TIMEOUT) {
     res.failure_reason = rtc::Optional<ProbeFailureReason>(kTimeout);
   } else {
     RTC_NOTREACHED();
   }
 
   return res;
 }
 
+ParsedRtcEventLog::BweAckedBitrateEvent ParsedRtcEventLog::GetAckedBitrate(
+    size_t index) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::BWE_ACKED_BITRATE_EVENT);
+  RTC_CHECK(event.has_acked_bitrate());
+  const rtclog::BweAckedBitrate& ab_event = event.acked_bitrate();
+  BweAckedBitrateEvent res;
+  res.timestamp = GetTimestamp(index);
+  RTC_CHECK(ab_event.has_bitrate_bps());
+  res.bitrate_bps = ab_event.bitrate_bps();
+
+  return res;
+}
+
+ParsedRtcEventLog::AlrStateEvent ParsedRtcEventLog::GetAlrState(
+    size_t index) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::ALR_STATE_EVENT);
+  RTC_CHECK(event.has_alr_state());
+  const rtclog::AlrState& alr_event = event.alr_state();
+  AlrStateEvent res;
+  res.timestamp = GetTimestamp(index);
+  RTC_CHECK(alr_event.has_in_alr());
+  res.in_alr = alr_event.in_alr();
+  RTC_CHECK(alr_event.has_usage_bps());
+  res.usage_bps = alr_event.usage_bps();
+
+  return res;
+}
+
+ParsedRtcEventLog::PacketQueueTime ParsedRtcEventLog::GetQueueTime(
+    size_t index) const {
+  RTC_CHECK_LT(index, GetNumberOfEvents());
+  const rtclog::Event& event = events_[index];
+  RTC_CHECK(event.has_type());
+  RTC_CHECK_EQ(event.type(), rtclog::Event::PACKET_QUEUE_TIME);
+  RTC_CHECK(event.has_packet_queue_time());
+  const rtclog::PacketQueueTime& queue_event = event.packet_queue_time();
+  PacketQueueTime res;
+  res.timestamp = GetTimestamp(index);
+  RTC_CHECK(queue_event.has_queue_time_ms());
+  res.queue_time_ms = queue_event.queue_time_ms();
+  RTC_CHECK(queue_event.has_ssrc());
+  res.ssrc = queue_event.ssrc();
+  return res;
+}
+
 // Returns the MediaType for registered SSRCs. Search from the end to use last
 // registered types first.
 ParsedRtcEventLog::MediaType ParsedRtcEventLog::GetMediaType(
     uint32_t ssrc,
     PacketDirection direction) const {
   for (auto rit = streams_.rbegin(); rit != streams_.rend(); ++rit) {
     if (rit->ssrc == ssrc && rit->direction == direction)
       return rit->media_type;
   }
   return MediaType::ANY;
 }
 }  // namespace webrtc