Rename some variables and methods in RTC event log.
Rename loss based and delay based bwe updates in proto (and correspondingly in the C++ code).
BUG=webrtc:6423
Review-Url: https://codereview.webrtc.org/2705613002
Cr-Commit-Position: refs/heads/master@{#16719}
diff --git a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
index a47aa65..7366c29 100644
--- a/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h
@@ -54,13 +54,13 @@
MOCK_METHOD1(LogAudioPlayout, void(uint32_t ssrc));
- MOCK_METHOD3(LogBwePacketLossEvent,
- void(int32_t bitrate,
+ MOCK_METHOD3(LogLossBasedBweUpdate,
+ void(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets));
- MOCK_METHOD2(LogBwePacketDelayEvent,
- void(int32_t bitrate, BandwidthUsage detector_state));
+ MOCK_METHOD2(LogDelayBasedBweUpdate,
+ void(int32_t bitrate_bps, BandwidthUsage detector_state));
MOCK_METHOD1(LogAudioNetworkAdaptation,
void(const AudioNetworkAdaptor::EncoderRuntimeConfig& config));
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.cc b/webrtc/logging/rtc_event_log/rtc_event_log.cc
index 96f1ea1..88f6b3a 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.cc
@@ -74,10 +74,10 @@
const uint8_t* packet,
size_t length) override;
void LogAudioPlayout(uint32_t ssrc) override;
- void LogBwePacketLossEvent(int32_t bitrate,
+ void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) override;
- void LogBwePacketDelayEvent(int32_t bitrate,
+ void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) override;
void LogAudioNetworkAdaptation(
const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override;
@@ -131,18 +131,18 @@
return rtclog::ANY;
}
-rtclog::BwePacketDelayEvent::DetectorState ConvertDetectorState(
+rtclog::DelayBasedBweUpdate::DetectorState ConvertDetectorState(
BandwidthUsage state) {
switch (state) {
case BandwidthUsage::kBwNormal:
- return rtclog::BwePacketDelayEvent::BWE_NORMAL;
+ return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
case BandwidthUsage::kBwUnderusing:
- return rtclog::BwePacketDelayEvent::BWE_UNDERUSING;
+ return rtclog::DelayBasedBweUpdate::BWE_UNDERUSING;
case BandwidthUsage::kBwOverusing:
- return rtclog::BwePacketDelayEvent::BWE_OVERUSING;
+ return rtclog::DelayBasedBweUpdate::BWE_OVERUSING;
}
RTC_NOTREACHED();
- return rtclog::BwePacketDelayEvent::BWE_NORMAL;
+ return rtclog::DelayBasedBweUpdate::BWE_NORMAL;
}
// The RTP and RTCP buffers reserve space for twice the expected number of
@@ -439,26 +439,26 @@
StoreEvent(&event);
}
-void RtcEventLogImpl::LogBwePacketLossEvent(int32_t bitrate,
+void RtcEventLogImpl::LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(rtc::TimeMicros());
- event->set_type(rtclog::Event::BWE_PACKET_LOSS_EVENT);
- auto bwe_event = event->mutable_bwe_packet_loss_event();
- bwe_event->set_bitrate(bitrate);
+ event->set_type(rtclog::Event::LOSS_BASED_BWE_UPDATE);
+ auto bwe_event = event->mutable_loss_based_bwe_update();
+ bwe_event->set_bitrate_bps(bitrate_bps);
bwe_event->set_fraction_loss(fraction_loss);
bwe_event->set_total_packets(total_packets);
StoreEvent(&event);
}
-void RtcEventLogImpl::LogBwePacketDelayEvent(int32_t bitrate,
+void RtcEventLogImpl::LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) {
std::unique_ptr<rtclog::Event> event(new rtclog::Event());
event->set_timestamp_us(rtc::TimeMicros());
- event->set_type(rtclog::Event::BWE_PACKET_DELAY_EVENT);
- auto bwe_event = event->mutable_bwe_packet_delay_event();
- bwe_event->set_bitrate(bitrate);
+ event->set_type(rtclog::Event::DELAY_BASED_BWE_UPDATE);
+ auto bwe_event = event->mutable_delay_based_bwe_update();
+ bwe_event->set_bitrate_bps(bitrate_bps);
bwe_event->set_detector_state(ConvertDetectorState(detector_state));
StoreEvent(&event);
}
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.h b/webrtc/logging/rtc_event_log/rtc_event_log.h
index 766fd89..f1bbcbb 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.h
@@ -112,12 +112,12 @@
virtual void LogAudioPlayout(uint32_t ssrc) = 0;
// Logs a bitrate update from the bandwidth estimator based on packet loss.
- virtual void LogBwePacketLossEvent(int32_t bitrate,
+ virtual void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) = 0;
// Logs a bitrate update from the bandwidth estimator based on delay changes.
- virtual void LogBwePacketDelayEvent(int32_t bitrate,
+ virtual void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) = 0;
// Logs audio encoder re-configuration driven by audio network adaptor.
@@ -162,10 +162,10 @@
const uint8_t* packet,
size_t length) override {}
void LogAudioPlayout(uint32_t ssrc) override {}
- void LogBwePacketLossEvent(int32_t bitrate,
+ void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) override {}
- void LogBwePacketDelayEvent(int32_t bitrate,
+ void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) override {}
void LogAudioNetworkAdaptation(
const AudioNetworkAdaptor::EncoderRuntimeConfig& config) override {}
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log.proto b/webrtc/logging/rtc_event_log/rtc_event_log.proto
index 0da910a..8f654f9 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log.proto
+++ b/webrtc/logging/rtc_event_log/rtc_event_log.proto
@@ -31,8 +31,8 @@
RTP_EVENT = 3;
RTCP_EVENT = 4;
AUDIO_PLAYOUT_EVENT = 5;
- BWE_PACKET_LOSS_EVENT = 6;
- BWE_PACKET_DELAY_EVENT = 7;
+ LOSS_BASED_BWE_UPDATE = 6;
+ DELAY_BASED_BWE_UPDATE = 7;
VIDEO_RECEIVER_CONFIG_EVENT = 8;
VIDEO_SENDER_CONFIG_EVENT = 9;
AUDIO_RECEIVER_CONFIG_EVENT = 10;
@@ -52,11 +52,11 @@
// optional - but required if type == AUDIO_PLAYOUT_EVENT
optional AudioPlayoutEvent audio_playout_event = 5;
- // optional - but required if type == BWE_PACKET_LOSS_EVENT
- optional BwePacketLossEvent bwe_packet_loss_event = 6;
+ // optional - but required if type == LOSS_BASED_BWE_UPDATE
+ optional LossBasedBweUpdate loss_based_bwe_update = 6;
- // optional - but required if type == BWE_PACKET_DELAY_EVENT
- optional BwePacketDelayEvent bwe_packet_delay_event = 7;
+ // optional - but required if type == DELAY_BASED_BWE_UPDATE
+ optional DelayBasedBweUpdate delay_based_bwe_update = 7;
// optional - but required if type == VIDEO_RECEIVER_CONFIG_EVENT
optional VideoReceiveConfig video_receiver_config = 8;
@@ -106,9 +106,9 @@
optional uint32 local_ssrc = 2;
}
-message BwePacketLossEvent {
+message LossBasedBweUpdate {
// required - Bandwidth estimate (in bps) after the update.
- optional int32 bitrate = 1;
+ optional int32 bitrate_bps = 1;
// required - Fraction of lost packets since last receiver report
// computed as floor( 256 * (#lost_packets / #total_packets) ).
@@ -120,7 +120,7 @@
optional int32 total_packets = 3;
}
-message BwePacketDelayEvent {
+message DelayBasedBweUpdate {
enum DetectorState {
BWE_NORMAL = 0;
BWE_UNDERUSING = 1;
@@ -128,7 +128,7 @@
}
// required - Bandwidth estimate (in bps) after the update.
- optional int32 bitrate = 1;
+ optional int32 bitrate_bps = 1;
// required - The state of the overuse detector.
optional DetectorState detector_state = 2;
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 012b7e1..713d4fc 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.cc
@@ -69,10 +69,10 @@
return ParsedRtcEventLog::EventType::RTCP_EVENT;
case rtclog::Event::AUDIO_PLAYOUT_EVENT:
return ParsedRtcEventLog::EventType::AUDIO_PLAYOUT_EVENT;
- case rtclog::Event::BWE_PACKET_LOSS_EVENT:
- return ParsedRtcEventLog::EventType::BWE_PACKET_LOSS_EVENT;
- case rtclog::Event::BWE_PACKET_DELAY_EVENT:
- return ParsedRtcEventLog::EventType::BWE_PACKET_DELAY_EVENT;
+ case rtclog::Event::LOSS_BASED_BWE_UPDATE:
+ return ParsedRtcEventLog::EventType::LOSS_BASED_BWE_UPDATE;
+ case rtclog::Event::DELAY_BASED_BWE_UPDATE:
+ return ParsedRtcEventLog::EventType::DELAY_BASED_BWE_UPDATE;
case rtclog::Event::VIDEO_RECEIVER_CONFIG_EVENT:
return ParsedRtcEventLog::EventType::VIDEO_RECEIVER_CONFIG_EVENT;
case rtclog::Event::VIDEO_SENDER_CONFIG_EVENT:
@@ -89,13 +89,13 @@
}
BandwidthUsage GetRuntimeDetectorState(
- rtclog::BwePacketDelayEvent::DetectorState detector_state) {
+ rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
switch (detector_state) {
- case rtclog::BwePacketDelayEvent::BWE_NORMAL:
+ case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
return kBwNormal;
- case rtclog::BwePacketDelayEvent::BWE_UNDERUSING:
+ case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
return kBwUnderusing;
- case rtclog::BwePacketDelayEvent::BWE_OVERUSING:
+ case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
return kBwOverusing;
}
RTC_NOTREACHED();
@@ -461,19 +461,19 @@
}
}
-void ParsedRtcEventLog::GetBwePacketLossEvent(size_t index,
- int32_t* bitrate,
+void ParsedRtcEventLog::GetLossBasedBweUpdate(size_t index,
+ int32_t* bitrate_bps,
uint8_t* fraction_loss,
int32_t* total_packets) 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_PACKET_LOSS_EVENT);
- RTC_CHECK(event.has_bwe_packet_loss_event());
- const rtclog::BwePacketLossEvent& loss_event = event.bwe_packet_loss_event();
- RTC_CHECK(loss_event.has_bitrate());
- if (bitrate != nullptr) {
- *bitrate = loss_event.bitrate();
+ RTC_CHECK_EQ(event.type(), rtclog::Event::LOSS_BASED_BWE_UPDATE);
+ RTC_CHECK(event.has_loss_based_bwe_update());
+ const rtclog::LossBasedBweUpdate& loss_event = event.loss_based_bwe_update();
+ RTC_CHECK(loss_event.has_bitrate_bps());
+ if (bitrate_bps != nullptr) {
+ *bitrate_bps = loss_event.bitrate_bps();
}
RTC_CHECK(loss_event.has_fraction_loss());
if (fraction_loss != nullptr) {
@@ -485,20 +485,20 @@
}
}
-void ParsedRtcEventLog::GetBwePacketDelayEvent(
+void ParsedRtcEventLog::GetDelayBasedBweUpdate(
size_t index,
- int32_t* bitrate,
+ int32_t* bitrate_bps,
BandwidthUsage* detector_state) 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_PACKET_DELAY_EVENT);
- RTC_CHECK(event.has_bwe_packet_delay_event());
- const rtclog::BwePacketDelayEvent& delay_event =
- event.bwe_packet_delay_event();
- RTC_CHECK(delay_event.has_bitrate());
- if (bitrate != nullptr) {
- *bitrate = delay_event.bitrate();
+ RTC_CHECK_EQ(event.type(), rtclog::Event::DELAY_BASED_BWE_UPDATE);
+ RTC_CHECK(event.has_delay_based_bwe_update());
+ const rtclog::DelayBasedBweUpdate& delay_event =
+ event.delay_based_bwe_update();
+ RTC_CHECK(delay_event.has_bitrate_bps());
+ if (bitrate_bps != nullptr) {
+ *bitrate_bps = delay_event.bitrate_bps();
}
RTC_CHECK(delay_event.has_detector_state());
if (detector_state != nullptr) {
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
index c81b8fb..739ccee 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_parser.h
@@ -42,8 +42,8 @@
RTP_EVENT = 3,
RTCP_EVENT = 4,
AUDIO_PLAYOUT_EVENT = 5,
- BWE_PACKET_LOSS_EVENT = 6,
- BWE_PACKET_DELAY_EVENT = 7,
+ LOSS_BASED_BWE_UPDATE = 6,
+ DELAY_BASED_BWE_UPDATE = 7,
VIDEO_RECEIVER_CONFIG_EVENT = 8,
VIDEO_SENDER_CONFIG_EVENT = 9,
AUDIO_RECEIVER_CONFIG_EVENT = 10,
@@ -120,8 +120,8 @@
// the corresponding output parameters. Each output parameter can be set to
// nullptr if that
// value isn't needed.
- void GetBwePacketLossEvent(size_t index,
- int32_t* bitrate,
+ void GetLossBasedBweUpdate(size_t index,
+ int32_t* bitrate_bps,
uint8_t* fraction_loss,
int32_t* total_packets) const;
@@ -129,8 +129,8 @@
// and stores the values in the corresponding output parameters. Each output
// parameter can be set to nullptr if that
// value isn't needed.
- void GetBwePacketDelayEvent(size_t index,
- int32_t* bitrate,
+ void GetDelayBasedBweUpdate(size_t index,
+ int32_t* bitrate_bps,
BandwidthUsage* detector_state) const;
// Reads a audio network adaptation event to a (non-NULL)
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
index 1173bf5..ae264aa 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest.cc
@@ -283,7 +283,7 @@
for (size_t i = 0; i < playout_count; i++) {
playout_ssrcs.push_back(prng.Rand<uint32_t>());
}
- // Create bwe_loss_count random bitrate updates for BwePacketLoss.
+ // Create bwe_loss_count random bitrate updates for LossBasedBwe.
for (size_t i = 0; i < bwe_loss_count; i++) {
bwe_loss_updates.push_back(
std::make_pair(prng.Rand<int32_t>(), prng.Rand<uint8_t>()));
@@ -333,7 +333,7 @@
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
}
if (i * bwe_loss_count >= bwe_loss_index * rtp_count) {
- log_dumper->LogBwePacketLossEvent(
+ log_dumper->LogLossBasedBweUpdate(
bwe_loss_updates[bwe_loss_index - 1].first,
bwe_loss_updates[bwe_loss_index - 1].second, i);
bwe_loss_index++;
@@ -500,7 +500,7 @@
remove(temp_filename.c_str());
}
-TEST(RtcEventLogTest, LogPacketLossEventAndReadBack) {
+TEST(RtcEventLogTest, LogLossBasedBweUpdateAndReadBack) {
Random prng(1234);
// Generate a random packet loss event.
@@ -520,7 +520,7 @@
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->StartLogging(temp_filename, 10000000);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogBwePacketLossEvent(bitrate, fraction_lost, total_packets);
+ log_dumper->LogLossBasedBweUpdate(bitrate, fraction_lost, total_packets);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StopLogging();
@@ -540,7 +540,7 @@
remove(temp_filename.c_str());
}
-TEST(RtcEventLogTest, LogPacketDelayEventAndReadBack) {
+TEST(RtcEventLogTest, LogDelayBasedBweUpdateAndReadBack) {
Random prng(1234);
// Generate 3 random packet delay event.
@@ -560,11 +560,11 @@
std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
log_dumper->StartLogging(temp_filename, 10000000);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogBwePacketDelayEvent(bitrate1, kBwNormal);
+ log_dumper->LogDelayBasedBweUpdate(bitrate1, kBwNormal);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogBwePacketDelayEvent(bitrate2, kBwOverusing);
+ log_dumper->LogDelayBasedBweUpdate(bitrate2, kBwOverusing);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
- log_dumper->LogBwePacketDelayEvent(bitrate3, kBwUnderusing);
+ log_dumper->LogDelayBasedBweUpdate(bitrate3, kBwUnderusing);
fake_clock.AdvanceTimeMicros(prng.Rand(1, 1000));
log_dumper->StopLogging();
diff --git a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
index 6d92b46..e7db593 100644
--- a/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
+++ b/webrtc/logging/rtc_event_log/rtc_event_log_unittest_helper.cc
@@ -44,13 +44,13 @@
}
BandwidthUsage GetRuntimeDetectorState(
- rtclog::BwePacketDelayEvent::DetectorState detector_state) {
+ rtclog::DelayBasedBweUpdate::DetectorState detector_state) {
switch (detector_state) {
- case rtclog::BwePacketDelayEvent::BWE_NORMAL:
+ case rtclog::DelayBasedBweUpdate::BWE_NORMAL:
return kBwNormal;
- case rtclog::BwePacketDelayEvent::BWE_UNDERUSING:
+ case rtclog::DelayBasedBweUpdate::BWE_UNDERUSING:
return kBwUnderusing;
- case rtclog::BwePacketDelayEvent::BWE_OVERUSING:
+ case rtclog::DelayBasedBweUpdate::BWE_OVERUSING:
return kBwOverusing;
}
RTC_NOTREACHED();
@@ -78,18 +78,18 @@
<< "Event of type " << type << " has "
<< (event.has_rtcp_packet() ? "" : "no ") << "RTCP packet";
}
- if ((type == rtclog::Event::BWE_PACKET_LOSS_EVENT) !=
- event.has_bwe_packet_loss_event()) {
+ if ((type == rtclog::Event::LOSS_BASED_BWE_UPDATE) !=
+ event.has_loss_based_bwe_update()) {
return ::testing::AssertionFailure()
<< "Event of type " << type << " has "
- << (event.has_bwe_packet_loss_event() ? "" : "no ") << "packet loss";
+ << (event.has_loss_based_bwe_update() ? "" : "no ") << "loss update";
}
- if ((type == rtclog::Event::BWE_PACKET_DELAY_EVENT) !=
- event.has_bwe_packet_delay_event()) {
+ if ((type == rtclog::Event::DELAY_BASED_BWE_UPDATE) !=
+ event.has_delay_based_bwe_update()) {
return ::testing::AssertionFailure()
<< "Event of type " << type << " has "
- << (event.has_bwe_packet_delay_event() ? "" : "no ")
- << "packet delay";
+ << (event.has_delay_based_bwe_update() ? "" : "no ")
+ << "delay update";
}
if ((type == rtclog::Event::AUDIO_PLAYOUT_EVENT) !=
event.has_audio_playout_event()) {
@@ -475,10 +475,10 @@
int32_t total_packets) {
const rtclog::Event& event = parsed_log.events_[index];
ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::BWE_PACKET_LOSS_EVENT, event.type());
- const rtclog::BwePacketLossEvent& bwe_event = event.bwe_packet_loss_event();
- ASSERT_TRUE(bwe_event.has_bitrate());
- EXPECT_EQ(bitrate, bwe_event.bitrate());
+ ASSERT_EQ(rtclog::Event::LOSS_BASED_BWE_UPDATE, event.type());
+ const rtclog::LossBasedBweUpdate& bwe_event = event.loss_based_bwe_update();
+ ASSERT_TRUE(bwe_event.has_bitrate_bps());
+ EXPECT_EQ(bitrate, bwe_event.bitrate_bps());
ASSERT_TRUE(bwe_event.has_fraction_loss());
EXPECT_EQ(fraction_loss, bwe_event.fraction_loss());
ASSERT_TRUE(bwe_event.has_total_packets());
@@ -488,7 +488,7 @@
int32_t parsed_bitrate;
uint8_t parsed_fraction_loss;
int32_t parsed_total_packets;
- parsed_log.GetBwePacketLossEvent(
+ parsed_log.GetLossBasedBweUpdate(
index, &parsed_bitrate, &parsed_fraction_loss, &parsed_total_packets);
EXPECT_EQ(bitrate, parsed_bitrate);
EXPECT_EQ(fraction_loss, parsed_fraction_loss);
@@ -502,10 +502,10 @@
BandwidthUsage detector_state) {
const rtclog::Event& event = parsed_log.events_[index];
ASSERT_TRUE(IsValidBasicEvent(event));
- ASSERT_EQ(rtclog::Event::BWE_PACKET_DELAY_EVENT, event.type());
- const rtclog::BwePacketDelayEvent& bwe_event = event.bwe_packet_delay_event();
- ASSERT_TRUE(bwe_event.has_bitrate());
- EXPECT_EQ(bitrate, bwe_event.bitrate());
+ ASSERT_EQ(rtclog::Event::DELAY_BASED_BWE_UPDATE, event.type());
+ const rtclog::DelayBasedBweUpdate& bwe_event = event.delay_based_bwe_update();
+ ASSERT_TRUE(bwe_event.has_bitrate_bps());
+ EXPECT_EQ(bitrate, bwe_event.bitrate_bps());
ASSERT_TRUE(bwe_event.has_detector_state());
EXPECT_EQ(detector_state,
GetRuntimeDetectorState(bwe_event.detector_state()));
@@ -513,7 +513,7 @@
// Check consistency of the parser.
int32_t parsed_bitrate;
BandwidthUsage parsed_detector_state;
- parsed_log.GetBwePacketDelayEvent(index, &parsed_bitrate,
+ parsed_log.GetDelayBasedBweUpdate(index, &parsed_bitrate,
&parsed_detector_state);
EXPECT_EQ(bitrate, parsed_bitrate);
EXPECT_EQ(detector_state, parsed_detector_state);
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index ad8a6f9..2c683fd 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -286,7 +286,7 @@
last_fraction_loss_ != last_logged_fraction_loss_ ||
last_rtc_event_log_ms_ == -1 ||
now_ms - last_rtc_event_log_ms_ > kRtcEventLogPeriodMs) {
- event_log_->LogBwePacketLossEvent(capped_bitrate, last_fraction_loss_,
+ event_log_->LogLossBasedBweUpdate(capped_bitrate, last_fraction_loss_,
expected_packets_since_last_loss_update_);
last_logged_fraction_loss_ = last_fraction_loss_;
last_rtc_event_log_ms_ = now_ms;
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
index 7841f5f..825828b 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation_unittest.cc
@@ -66,7 +66,7 @@
TEST(SendSideBweTest, DoesntReapplyBitrateDecreaseWithoutFollowingRemb) {
MockRtcEventLog event_log;
EXPECT_CALL(event_log,
- LogBwePacketLossEvent(testing::Gt(0), testing::Gt(0), 0))
+ LogLossBasedBweUpdate(testing::Gt(0), testing::Gt(0), 0))
.Times(1);
SendSideBandwidthEstimation bwe(&event_log);
static const int kMinBitrateBps = 100000;
diff --git a/webrtc/modules/congestion_controller/delay_based_bwe.cc b/webrtc/modules/congestion_controller/delay_based_bwe.cc
index d18847f..71fefac 100644
--- a/webrtc/modules/congestion_controller/delay_based_bwe.cc
+++ b/webrtc/modules/congestion_controller/delay_based_bwe.cc
@@ -395,7 +395,7 @@
result.target_bitrate_bps);
if (event_log_ && (result.target_bitrate_bps != last_logged_bitrate_ ||
detector_.State() != last_logged_state_)) {
- event_log_->LogBwePacketDelayEvent(result.target_bitrate_bps,
+ event_log_->LogDelayBasedBweUpdate(result.target_bitrate_bps,
detector_.State());
last_logged_bitrate_ = result.target_bitrate_bps;
last_logged_state_ = detector_.State();
diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 2f8b7ed..5c0433a 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -435,24 +435,24 @@
case ParsedRtcEventLog::LOG_END: {
break;
}
- case ParsedRtcEventLog::BWE_PACKET_LOSS_EVENT: {
- BwePacketLossEvent bwe_update;
+ case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
+ break;
+ }
+ case ParsedRtcEventLog::LOSS_BASED_BWE_UPDATE: {
+ LossBasedBweUpdate bwe_update;
bwe_update.timestamp = parsed_log_.GetTimestamp(i);
- parsed_log_.GetBwePacketLossEvent(i, &bwe_update.new_bitrate,
- &bwe_update.fraction_loss,
- &bwe_update.expected_packets);
+ parsed_log_.GetLossBasedBweUpdate(i, &bwe_update.new_bitrate,
+ &bwe_update.fraction_loss,
+ &bwe_update.expected_packets);
bwe_loss_updates_.push_back(bwe_update);
break;
}
+ case ParsedRtcEventLog::DELAY_BASED_BWE_UPDATE: {
+ break;
+ }
case ParsedRtcEventLog::AUDIO_NETWORK_ADAPTATION_EVENT: {
break;
}
- case ParsedRtcEventLog::BWE_PACKET_DELAY_EVENT: {
- break;
- }
- case ParsedRtcEventLog::AUDIO_PLAYOUT_EVENT: {
- break;
- }
case ParsedRtcEventLog::UNKNOWN_EVENT: {
break;
}
diff --git a/webrtc/tools/event_log_visualizer/analyzer.h b/webrtc/tools/event_log_visualizer/analyzer.h
index bb7667f..f0557a2 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.h
+++ b/webrtc/tools/event_log_visualizer/analyzer.h
@@ -45,7 +45,7 @@
std::unique_ptr<rtcp::RtcpPacket> packet;
};
-struct BwePacketLossEvent {
+struct LossBasedBweUpdate {
uint64_t timestamp;
int32_t new_bitrate;
uint8_t fraction_loss;
@@ -150,7 +150,7 @@
std::map<StreamId, std::vector<LoggedRtcpPacket>> rtcp_packets_;
// A list of all updates from the send-side loss-based bandwidth estimator.
- std::vector<BwePacketLossEvent> bwe_loss_updates_;
+ std::vector<LossBasedBweUpdate> bwe_loss_updates_;
// Window and step size used for calculating moving averages, e.g. bitrate.
// The generated data points will be |step_| microseconds apart.
diff --git a/webrtc/voice_engine/channel.cc b/webrtc/voice_engine/channel.cc
index b0aa654..acd8052 100644
--- a/webrtc/voice_engine/channel.cc
+++ b/webrtc/voice_engine/channel.cc
@@ -130,20 +130,21 @@
}
}
- void LogBwePacketLossEvent(int32_t bitrate,
+ void LogLossBasedBweUpdate(int32_t bitrate_bps,
uint8_t fraction_loss,
int32_t total_packets) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
- event_log_->LogBwePacketLossEvent(bitrate, fraction_loss, total_packets);
+ event_log_->LogLossBasedBweUpdate(bitrate_bps, fraction_loss,
+ total_packets);
}
}
- void LogBwePacketDelayEvent(int32_t bitrate,
+ void LogDelayBasedBweUpdate(int32_t bitrate_bps,
BandwidthUsage detector_state) override {
rtc::CritScope lock(&crit_);
if (event_log_) {
- event_log_->LogBwePacketDelayEvent(bitrate, detector_state);
+ event_log_->LogDelayBasedBweUpdate(bitrate_bps, detector_state);
}
}