Add UMA for send bwe and pacer bitrate.
Review URL: https://codereview.webrtc.org/1434403004
Cr-Commit-Position: refs/heads/master@{#10675}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 128a992..326c1ba 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -114,7 +114,8 @@
return nullptr;
}
- void UpdateHistograms();
+ void UpdateSendHistograms();
+ void UpdateReceiveHistograms();
const Clock* const clock_;
@@ -148,14 +149,24 @@
RtcEventLog* event_log_ = nullptr;
- // The RateTrackers are only accessed (exclusively) from DeliverRtp or
- // DeliverRtcp, and from the destructor, and therefore doesn't need any
- // explicit synchronization.
+ // The following members are only accessed (exclusively) from one thread and
+ // from the destructor, and therefore doesn't need any explicit
+ // synchronization.
rtc::RateTracker received_video_bytes_per_sec_;
rtc::RateTracker received_audio_bytes_per_sec_;
rtc::RateTracker received_rtcp_bytes_per_sec_;
+ int64_t first_packet_sent_ms_;
int64_t first_rtp_packet_received_ms_;
+ // TODO(holmer): Remove this lock once BitrateController no longer calls
+ // OnNetworkChanged from multiple threads.
+ rtc::CriticalSection bitrate_crit_;
+ rtc::RateTracker estimated_send_bitrate_kbps_ GUARDED_BY(&bitrate_crit_);
+ rtc::RateTracker pacer_bitrate_kbps_ GUARDED_BY(&bitrate_crit_);
+ uint32_t target_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
+ uint32_t pacer_bitrate_bps_ GUARDED_BY(&bitrate_crit_);
+ int64_t last_bitrate_update_ms_ GUARDED_BY(&bitrate_crit_);
+
const rtc::scoped_ptr<CongestionController> congestion_controller_;
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
@@ -181,9 +192,17 @@
received_video_bytes_per_sec_(1000, 1),
received_audio_bytes_per_sec_(1000, 1),
received_rtcp_bytes_per_sec_(1000, 1),
+ first_packet_sent_ms_(-1),
first_rtp_packet_received_ms_(-1),
- congestion_controller_(new CongestionController(
- module_process_thread_.get(), call_stats_.get(), this)) {
+ estimated_send_bitrate_kbps_(1000, 1),
+ pacer_bitrate_kbps_(1000, 1),
+ target_bitrate_bps_(0),
+ pacer_bitrate_bps_(0),
+ last_bitrate_update_ms_(-1),
+ congestion_controller_(
+ new CongestionController(module_process_thread_.get(),
+ call_stats_.get(),
+ this)) {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
@@ -211,7 +230,8 @@
Call::~Call() {
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
- UpdateHistograms();
+ UpdateSendHistograms();
+ UpdateReceiveHistograms();
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
RTC_CHECK(video_send_streams_.empty());
@@ -224,7 +244,27 @@
Trace::ReturnTrace();
}
-void Call::UpdateHistograms() {
+void Call::UpdateSendHistograms() {
+ if (first_packet_sent_ms_ == -1)
+ return;
+ int64_t elapsed_sec =
+ (clock_->TimeInMilliseconds() - first_packet_sent_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+ return;
+ rtc::CritScope lock(&bitrate_crit_);
+ int send_bitrate_kbps = estimated_send_bitrate_kbps_.ComputeTotalRate();
+ int pacer_bitrate_kbps = pacer_bitrate_kbps_.ComputeTotalRate();
+ if (send_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.EstimatedSendBitrateInKbps",
+ send_bitrate_kbps);
+ }
+ if (pacer_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.PacerBitrateInKbps",
+ pacer_bitrate_kbps);
+ }
+}
+
+void Call::UpdateReceiveHistograms() {
if (first_rtp_packet_received_ms_ == -1)
return;
int64_t elapsed_sec =
@@ -529,11 +569,26 @@
}
void Call::OnSentPacket(const rtc::SentPacket& sent_packet) {
+ if (first_packet_sent_ms_ == -1)
+ first_packet_sent_ms_ = clock_->TimeInMilliseconds();
congestion_controller_->OnSentPacket(sent_packet);
}
void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
int64_t rtt_ms) {
+ int64_t now_ms = clock_->TimeInMilliseconds();
+ int64_t time_since_last_update_ms = 0;
+ {
+ rtc::CritScope lock(&bitrate_crit_);
+ if (last_bitrate_update_ms_ >= 0)
+ time_since_last_update_ms = now_ms - last_bitrate_update_ms_;
+ estimated_send_bitrate_kbps_.AddSamples(
+ time_since_last_update_ms * (target_bitrate_bps_ / 1000) / 1000);
+ pacer_bitrate_kbps_.AddSamples(time_since_last_update_ms *
+ (pacer_bitrate_bps_ / 1000) / 1000);
+ target_bitrate_bps_ = target_bitrate_bps;
+ last_bitrate_update_ms_ = now_ms;
+ }
uint32_t allocated_bitrate_bps = bitrate_allocator_->OnNetworkChanged(
target_bitrate_bps, fraction_loss, rtt_ms);
@@ -552,6 +607,10 @@
// set the pacer bitrate to the maximum of the two.
uint32_t pacer_bitrate_bps =
std::max(target_bitrate_bps, allocated_bitrate_bps);
+ {
+ rtc::CritScope lock(&bitrate_crit_);
+ pacer_bitrate_bps_ = pacer_bitrate_bps;
+ }
congestion_controller_->UpdatePacerBitrate(
target_bitrate_bps / 1000,
PacedSender::kDefaultPaceMultiplier * pacer_bitrate_bps / 1000,
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index c7133d0..b3174aa 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1944,6 +1944,7 @@
RunBaseTest(&test, FakeNetworkPipe::Config());
// Delete the call for Call stats to be reported.
+ sender_call_.reset();
receiver_call_.reset();
// Verify that stats have been updated once.
@@ -1952,6 +1953,9 @@
EXPECT_EQ(1,
test::NumHistogramSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.BitrateReceivedInKbps"));
+ EXPECT_EQ(
+ 1, test::NumHistogramSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
+ EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.PacerBitrateInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsSentPerMinute"));
