Add receive bitrate UMA stats.
Review URL: https://codereview.webrtc.org/1440603002
Cr-Commit-Position: refs/heads/master@{#10605}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 1144506..bdff1e7 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -33,6 +33,7 @@
#include "webrtc/system_wrappers/include/cpu_info.h"
#include "webrtc/system_wrappers/include/critical_section_wrapper.h"
#include "webrtc/system_wrappers/include/logging.h"
+#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/rw_lock_wrapper.h"
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/video/video_receive_stream.h"
@@ -105,6 +106,10 @@
return nullptr;
}
+ void UpdateHistograms();
+
+ const Clock* const clock_;
+
const int num_cpu_cores_;
const rtc::scoped_ptr<ProcessThread> module_process_thread_;
const rtc::scoped_ptr<CallStats> call_stats_;
@@ -135,6 +140,14 @@
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.
+ rtc::RateTracker received_video_bytes_per_sec_;
+ rtc::RateTracker received_audio_bytes_per_sec_;
+ rtc::RateTracker received_rtcp_bytes_per_sec_;
+ int64_t first_rtp_packet_received_ms_;
+
RTC_DISALLOW_COPY_AND_ASSIGN(Call);
};
} // namespace internal
@@ -146,15 +159,21 @@
namespace internal {
Call::Call(const Call::Config& config)
- : num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
+ : clock_(Clock::GetRealTimeClock()),
+ num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
call_stats_(new CallStats()),
- congestion_controller_(new CongestionController(
- module_process_thread_.get(), call_stats_.get())),
+ congestion_controller_(
+ new CongestionController(module_process_thread_.get(),
+ call_stats_.get())),
config_(config),
network_enabled_(true),
receive_crit_(RWLockWrapper::CreateRWLock()),
- send_crit_(RWLockWrapper::CreateRWLock()) {
+ send_crit_(RWLockWrapper::CreateRWLock()),
+ received_video_bytes_per_sec_(1000, 1),
+ received_audio_bytes_per_sec_(1000, 1),
+ received_rtcp_bytes_per_sec_(1000, 1),
+ first_rtp_packet_received_ms_(-1) {
RTC_DCHECK_GE(config.bitrate_config.min_bitrate_bps, 0);
RTC_DCHECK_GE(config.bitrate_config.start_bitrate_bps,
config.bitrate_config.min_bitrate_bps);
@@ -180,6 +199,7 @@
}
Call::~Call() {
+ UpdateHistograms();
RTC_DCHECK(configuration_thread_checker_.CalledOnValidThread());
RTC_CHECK(audio_send_ssrcs_.empty());
RTC_CHECK(video_send_ssrcs_.empty());
@@ -193,6 +213,35 @@
Trace::ReturnTrace();
}
+void Call::UpdateHistograms() {
+ if (first_rtp_packet_received_ms_ == -1)
+ return;
+ int64_t elapsed_sec =
+ (clock_->TimeInMilliseconds() - first_rtp_packet_received_ms_) / 1000;
+ if (elapsed_sec < metrics::kMinRunTimeInSeconds)
+ return;
+ int audio_bitrate_kbps =
+ received_audio_bytes_per_sec_.ComputeTotalRate() * 8 / 1000;
+ int video_bitrate_kbps =
+ received_video_bytes_per_sec_.ComputeTotalRate() * 8 / 1000;
+ int rtcp_bitrate_bps = received_rtcp_bytes_per_sec_.ComputeTotalRate() * 8;
+ if (video_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
+ video_bitrate_kbps);
+ }
+ if (audio_bitrate_kbps > 0) {
+ RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
+ audio_bitrate_kbps);
+ }
+ if (rtcp_bitrate_bps > 0) {
+ RTC_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
+ rtcp_bitrate_bps);
+ }
+ RTC_HISTOGRAM_COUNTS_100000(
+ "WebRTC.Call.BitrateReceivedInKbps",
+ audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
+}
+
PacketReceiver* Call::Receiver() {
// TODO(solenberg): Some test cases in EndToEndTest use this from a different
// thread. Re-enable once that is fixed.
@@ -527,6 +576,7 @@
// Do NOT broadcast! Also make sure it's a valid packet.
// Return DELIVERY_UNKNOWN_SSRC if it can be determined that
// there's no receiver of the packet.
+ received_rtcp_bytes_per_sec_.AddSamples(length);
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
@@ -559,12 +609,15 @@
if (length < 12)
return DELIVERY_PACKET_ERROR;
- uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
+ if (first_rtp_packet_received_ms_ == -1)
+ first_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
+ uint32_t ssrc = ByteReader<uint32_t>::ReadBigEndian(&packet[8]);
ReadLockScoped read_lock(*receive_crit_);
if (media_type == MediaType::ANY || media_type == MediaType::AUDIO) {
auto it = audio_receive_ssrcs_.find(ssrc);
if (it != audio_receive_ssrcs_.end()) {
+ received_audio_bytes_per_sec_.AddSamples(length);
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
@@ -576,6 +629,7 @@
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
auto it = video_receive_ssrcs_.find(ssrc);
if (it != video_receive_ssrcs_.end()) {
+ received_video_bytes_per_sec_.AddSamples(length);
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 2c89ab2..6ed3c47 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -1943,7 +1943,16 @@
test::ClearHistograms();
RunBaseTest(&test, FakeNetworkPipe::Config());
+ // Delete the call for Call stats to be reported.
+ receiver_call_.reset();
+
// Verify that stats have been updated once.
+ EXPECT_EQ(
+ 1, test::NumHistogramSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
+ EXPECT_EQ(1,
+ test::NumHistogramSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
+ EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Call.BitrateReceivedInKbps"));
+
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
