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Add UMA for send bwe and pacer bitrate.

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Review URL: https://codereview.webrtc.org/1434403004

Cr-Commit-Position: refs/heads/master@{#10675}
This commit is contained in:
stefan
2015-11-17 06:24:56 -08:00
committed by Commit bot
parent d9eec762ce
commit 18adf0a79d
2 changed files with 71 additions and 8 deletions
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75
webrtc/call/call.cc
View File

@ -114,7 +114,8 @@ class Call : public webrtc::Call, public PacketReceiver,
return nullptr;
}
void UpdateHistograms();
void UpdateSendHistograms();
void UpdateReceiveHistograms();
const Clock* const clock_;
@ -148,14 +149,24 @@ class Call : public webrtc::Call, public PacketReceiver,
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 @@ Call::Call(const Call::Config& config)
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(const Call::Config& config)
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 @@ Call::~Call() {
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::SignalNetworkState(NetworkState state) {
}
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 @@ void Call::OnNetworkChanged(uint32_t target_bitrate_bps, uint8_t fraction_loss,
// 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,

4
webrtc/video/end_to_end_tests.cc
View File

@ -1944,6 +1944,7 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {
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 @@ void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {
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"));