Remove RTC_LOGGED_* macro.
BUG=
Review-Url: https://codereview.webrtc.org/2326843003
Cr-Commit-Position: refs/heads/master@{#14174}
diff --git a/webrtc/modules/audio_device/audio_device_buffer.cc b/webrtc/modules/audio_device/audio_device_buffer.cc
index f87e068..b6c5df2 100644
--- a/webrtc/modules/audio_device/audio_device_buffer.cc
+++ b/webrtc/modules/audio_device/audio_device_buffer.cc
@@ -100,7 +100,7 @@
// I am avoiding use of the task queue here since we are under destruction
// and reading these members on the creating thread feels safe.
if (rec_callbacks_ > 0 && num_stat_reports_ > 0) {
- RTC_LOGGED_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros",
+ RTC_HISTOGRAM_BOOLEAN("WebRTC.Audio.RecordedOnlyZeros",
static_cast<int>(num_stat_reports_ == num_rec_level_is_zero_));
}
}
diff --git a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
index 7d9d5c3..523036d 100644
--- a/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
+++ b/webrtc/modules/bitrate_controller/send_side_bandwidth_estimation.cc
@@ -168,8 +168,8 @@
for (size_t i = 0; i < kNumUmaRampupMetrics; ++i) {
if (!rampup_uma_stats_updated_[i] &&
bitrate_kbps >= kUmaRampupMetrics[i].bitrate_kbps) {
- RTC_LOGGED_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
- now_ms - first_report_time_ms_);
+ RTC_HISTOGRAMS_COUNTS_100000(i, kUmaRampupMetrics[i].metric_name,
+ now_ms - first_report_time_ms_);
rampup_uma_stats_updated_[i] = true;
}
}
@@ -178,19 +178,19 @@
} else if (uma_update_state_ == kNoUpdate) {
uma_update_state_ = kFirstDone;
bitrate_at_2_seconds_kbps_ = bitrate_kbps;
- RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
- initially_lost_packets_, 0, 100, 50);
- RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt),
- 0, 2000, 50);
- RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
- bitrate_at_2_seconds_kbps_, 0, 2000, 50);
+ RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitiallyLostPackets",
+ initially_lost_packets_, 0, 100, 50);
+ RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialRtt", static_cast<int>(rtt), 0,
+ 2000, 50);
+ RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialBandwidthEstimate",
+ bitrate_at_2_seconds_kbps_, 0, 2000, 50);
} else if (uma_update_state_ == kFirstDone &&
now_ms - first_report_time_ms_ >= kBweConverganceTimeMs) {
uma_update_state_ = kDone;
int bitrate_diff_kbps =
std::max(bitrate_at_2_seconds_kbps_ - bitrate_kbps, 0);
- RTC_LOGGED_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff",
- bitrate_diff_kbps, 0, 2000, 50);
+ RTC_HISTOGRAM_COUNTS("WebRTC.BWE.InitialVsConvergedDiff", bitrate_diff_kbps,
+ 0, 2000, 50);
}
}
diff --git a/webrtc/modules/congestion_controller/delay_based_bwe.cc b/webrtc/modules/congestion_controller/delay_based_bwe.cc
index 86ef6f2..bd9f3ee 100644
--- a/webrtc/modules/congestion_controller/delay_based_bwe.cc
+++ b/webrtc/modules/congestion_controller/delay_based_bwe.cc
@@ -32,7 +32,6 @@
kAbsSendTimeFraction + kAbsSendTimeInterArrivalUpshift;
constexpr double kTimestampToMs =
1000.0 / static_cast<double>(1 << kInterArrivalShift);
-
// This ssrc is used to fulfill the current API but will be removed
// after the API has been changed.
constexpr uint32_t kFixedSsrc = 0;
@@ -59,9 +58,9 @@
const std::vector<PacketInfo>& packet_feedback_vector) {
RTC_DCHECK(network_thread_.CalledOnValidThread());
if (!uma_recorded_) {
- RTC_LOGGED_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
- BweNames::kSendSideTransportSeqNum,
- BweNames::kBweNamesMax);
+ RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
+ BweNames::kSendSideTransportSeqNum,
+ BweNames::kBweNamesMax);
uma_recorded_ = true;
}
for (const auto& packet_info : packet_feedback_vector) {
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
index bcd360d..786223f 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_abs_send_time.cc
@@ -239,9 +239,8 @@
uint32_t ssrc) {
RTC_CHECK(send_time_24bits < (1ul << 24));
if (!uma_recorded_) {
- RTC_LOGGED_HISTOGRAM_ENUMERATION(kBweTypeHistogram,
- BweNames::kReceiverAbsSendTime,
- BweNames::kBweNamesMax);
+ RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, BweNames::kReceiverAbsSendTime,
+ BweNames::kBweNamesMax);
uma_recorded_ = true;
}
// Shift up send time to use the full 32 bits that inter_arrival works with,
diff --git a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
index b1e478f..0115b14 100644
--- a/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
+++ b/webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator_single_stream.cc
@@ -77,8 +77,7 @@
BweNames type = BweNames::kReceiverTOffset;
if (!header.extension.hasTransmissionTimeOffset)
type = BweNames::kReceiverNoExtension;
- RTC_LOGGED_HISTOGRAM_ENUMERATION(
- kBweTypeHistogram, type, BweNames::kBweNamesMax);
+ RTC_HISTOGRAM_ENUMERATION(kBweTypeHistogram, type, BweNames::kBweNamesMax);
uma_recorded_ = true;
}
uint32_t ssrc = header.ssrc;
diff --git a/webrtc/modules/video_coding/jitter_buffer.cc b/webrtc/modules/video_coding/jitter_buffer.cc
index 02d8afa..345bdf7 100644
--- a/webrtc/modules/video_coding/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/jitter_buffer.cc
@@ -285,18 +285,18 @@
return;
}
- RTC_LOGGED_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
- num_discarded_packets_ * 100 / num_packets_);
- RTC_LOGGED_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
- num_duplicated_packets_ * 100 / num_packets_);
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
+ num_discarded_packets_ * 100 / num_packets_);
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
+ num_duplicated_packets_ * 100 / num_packets_);
int total_frames =
receive_statistics_.key_frames + receive_statistics_.delta_frames;
if (total_frames > 0) {
- RTC_LOGGED_HISTOGRAM_COUNTS_100(
+ RTC_HISTOGRAM_COUNTS_100(
"WebRTC.Video.CompleteFramesReceivedPerSecond",
static_cast<int>((total_frames / elapsed_sec) + 0.5f));
- RTC_LOGGED_HISTOGRAM_COUNTS_1000(
+ RTC_HISTOGRAM_COUNTS_1000(
"WebRTC.Video.KeyFramesReceivedInPermille",
static_cast<int>(
(receive_statistics_.key_frames * 1000.0f / total_frames) + 0.5f));
diff --git a/webrtc/modules/video_coding/timing.cc b/webrtc/modules/video_coding/timing.cc
index b20a18f..29b064e 100644
--- a/webrtc/modules/video_coding/timing.cc
+++ b/webrtc/modules/video_coding/timing.cc
@@ -63,14 +63,14 @@
if (elapsed_sec < metrics::kMinRunTimeInSeconds) {
return;
}
- RTC_LOGGED_HISTOGRAM_COUNTS_100(
+ RTC_HISTOGRAM_COUNTS_100(
"WebRTC.Video.DecodedFramesPerSecond",
static_cast<int>((num_decoded_frames_ / elapsed_sec) + 0.5f));
- RTC_LOGGED_HISTOGRAM_PERCENTAGE(
+ RTC_HISTOGRAM_PERCENTAGE(
"WebRTC.Video.DelayedFramesToRenderer",
num_delayed_decoded_frames_ * 100 / num_decoded_frames_);
if (num_delayed_decoded_frames_ > 0) {
- RTC_LOGGED_HISTOGRAM_COUNTS_1000(
+ RTC_HISTOGRAM_COUNTS_1000(
"WebRTC.Video.DelayedFramesToRenderer_AvgDelayInMs",
sum_missed_render_deadline_ms_ / num_delayed_decoded_frames_);
}