Use RateCounter for received bitrate stats:
"WebRTC.Call.BitrateReceivedInKbps"
"WebRTC.Call.VideoBitrateReceivedInKbps"
"WebRTC.Call.AudioBitrateReceivedInKbps"
"WebRTC.Call.RtcpBitrateReceivedInBps"
Reports the average of periodically computed bitrates over a call (2 sec interval). Adds possibility to later modify the stats and use/report the periodic bitrates.
BUG=webrtc:5283
Review-Url: https://codereview.webrtc.org/2303763002
Cr-Commit-Position: refs/heads/master@{#14119}
diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
index 5aa7228..4b5bd3a 100644
--- a/webrtc/call/call.cc
+++ b/webrtc/call/call.cc
@@ -43,6 +43,7 @@
#include "webrtc/system_wrappers/include/trace.h"
#include "webrtc/video/call_stats.h"
#include "webrtc/video/send_delay_stats.h"
+#include "webrtc/video/stats_counter.h"
#include "webrtc/video/video_receive_stream.h"
#include "webrtc/video/video_send_stream.h"
#include "webrtc/video/vie_remb.h"
@@ -176,12 +177,11 @@
// The following members are only accessed (exclusively) from one thread and
// from the destructor, and therefore doesn't need any explicit
// synchronization.
- int64_t received_video_bytes_;
- int64_t received_audio_bytes_;
- int64_t received_rtcp_bytes_;
- int64_t first_rtp_packet_received_ms_;
- int64_t last_rtp_packet_received_ms_;
int64_t first_packet_sent_ms_;
+ RateCounter received_bytes_per_second_counter_;
+ RateCounter received_audio_bytes_per_second_counter_;
+ RateCounter received_video_bytes_per_second_counter_;
+ RateCounter received_rtcp_bytes_per_second_counter_;
// TODO(holmer): Remove this lock once BitrateController no longer calls
// OnNetworkChanged from multiple threads.
@@ -239,12 +239,11 @@
receive_crit_(RWLockWrapper::CreateRWLock()),
send_crit_(RWLockWrapper::CreateRWLock()),
event_log_(RtcEventLog::Create(webrtc::Clock::GetRealTimeClock())),
- received_video_bytes_(0),
- received_audio_bytes_(0),
- received_rtcp_bytes_(0),
- first_rtp_packet_received_ms_(-1),
- last_rtp_packet_received_ms_(-1),
first_packet_sent_ms_(-1),
+ received_bytes_per_second_counter_(clock_, nullptr, true),
+ received_audio_bytes_per_second_counter_(clock_, nullptr, true),
+ received_video_bytes_per_second_counter_(clock_, nullptr, true),
+ received_rtcp_bytes_per_second_counter_(clock_, nullptr, true),
estimated_send_bitrate_sum_kbits_(0),
pacer_bitrate_sum_kbits_(0),
min_allocated_send_bitrate_bps_(0),
@@ -341,30 +340,31 @@
}
void Call::UpdateReceiveHistograms() {
- if (first_rtp_packet_received_ms_ == -1)
- return;
- int64_t elapsed_sec =
- (last_rtp_packet_received_ms_ - first_rtp_packet_received_ms_) / 1000;
- if (elapsed_sec < metrics::kMinRunTimeInSeconds)
- return;
- int audio_bitrate_kbps = received_audio_bytes_ * 8 / elapsed_sec / 1000;
- int video_bitrate_kbps = received_video_bytes_ * 8 / elapsed_sec / 1000;
- int rtcp_bitrate_bps = received_rtcp_bytes_ * 8 / elapsed_sec;
- if (video_bitrate_kbps > 0) {
+ const int kMinRequiredPeriodicSamples = 5;
+ AggregatedStats video_bytes_per_sec =
+ received_video_bytes_per_second_counter_.GetStats();
+ if (video_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.VideoBitrateReceivedInKbps",
- video_bitrate_kbps);
+ video_bytes_per_sec.average * 8 / 1000);
}
- if (audio_bitrate_kbps > 0) {
+ AggregatedStats audio_bytes_per_sec =
+ received_audio_bytes_per_second_counter_.GetStats();
+ if (audio_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.AudioBitrateReceivedInKbps",
- audio_bitrate_kbps);
+ audio_bytes_per_sec.average * 8 / 1000);
}
- if (rtcp_bitrate_bps > 0) {
+ AggregatedStats rtcp_bytes_per_sec =
+ received_rtcp_bytes_per_second_counter_.GetStats();
+ if (rtcp_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.RtcpBitrateReceivedInBps",
- rtcp_bitrate_bps);
+ rtcp_bytes_per_sec.average * 8);
}
- RTC_LOGGED_HISTOGRAM_COUNTS_100000(
- "WebRTC.Call.BitrateReceivedInKbps",
- audio_bitrate_kbps + video_bitrate_kbps + rtcp_bitrate_bps / 1000);
+ AggregatedStats recv_bytes_per_sec =
+ received_bytes_per_second_counter_.GetStats();
+ if (recv_bytes_per_sec.num_samples > kMinRequiredPeriodicSamples) {
+ RTC_LOGGED_HISTOGRAM_COUNTS_100000("WebRTC.Call.BitrateReceivedInKbps",
+ recv_bytes_per_sec.average * 8 / 1000);
+ }
}
PacketReceiver* Call::Receiver() {
@@ -843,7 +843,11 @@
// TODO(pbos): 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_ += length;
+ if (received_bytes_per_second_counter_.HasSample()) {
+ // First RTP packet has been received.
+ received_bytes_per_second_counter_.Add(static_cast<int>(length));
+ received_rtcp_bytes_per_second_counter_.Add(static_cast<int>(length));
+ }
bool rtcp_delivered = false;
if (media_type == MediaType::ANY || media_type == MediaType::VIDEO) {
ReadLockScoped read_lock(*receive_crit_);
@@ -889,16 +893,13 @@
if (length < 12)
return DELIVERY_PACKET_ERROR;
- last_rtp_packet_received_ms_ = clock_->TimeInMilliseconds();
- if (first_rtp_packet_received_ms_ == -1)
- first_rtp_packet_received_ms_ = last_rtp_packet_received_ms_;
-
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_ += length;
+ received_bytes_per_second_counter_.Add(static_cast<int>(length));
+ received_audio_bytes_per_second_counter_.Add(static_cast<int>(length));
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
@@ -910,7 +911,8 @@
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_ += length;
+ received_bytes_per_second_counter_.Add(static_cast<int>(length));
+ received_video_bytes_per_second_counter_.Add(static_cast<int>(length));
auto status = it->second->DeliverRtp(packet, length, packet_time)
? DELIVERY_OK
: DELIVERY_PACKET_ERROR;
diff --git a/webrtc/video/stats_counter.cc b/webrtc/video/stats_counter.cc
index 42d23ad..5bcef99 100644
--- a/webrtc/video/stats_counter.cc
+++ b/webrtc/video/stats_counter.cc
@@ -74,6 +74,10 @@
return aggregated_counter_->ComputeStats();
}
+bool StatsCounter::HasSample() const {
+ return last_process_time_ms_ != -1;
+}
+
bool StatsCounter::TimeToProcess() {
int64_t now = clock_->TimeInMilliseconds();
if (last_process_time_ms_ == -1)
diff --git a/webrtc/video/stats_counter.h b/webrtc/video/stats_counter.h
index c272b62..ba1bd45 100644
--- a/webrtc/video/stats_counter.h
+++ b/webrtc/video/stats_counter.h
@@ -82,6 +82,9 @@
AggregatedStats GetStats();
+ // Checks if a sample has been added (i.e. Add or Set called).
+ bool HasSample() const;
+
protected:
StatsCounter(Clock* clock,
bool include_empty_intervals,
diff --git a/webrtc/video/stats_counter_unittest.cc b/webrtc/video/stats_counter_unittest.cc
index 5dd8b72..d054eaa 100644
--- a/webrtc/video/stats_counter_unittest.cc
+++ b/webrtc/video/stats_counter_unittest.cc
@@ -72,6 +72,13 @@
EXPECT_EQ(1, observer->num_calls_);
}
+TEST_F(StatsCounterTest, HasSample) {
+ AvgCounter counter(&clock_, nullptr);
+ EXPECT_FALSE(counter.HasSample());
+ counter.Add(1);
+ EXPECT_TRUE(counter.HasSample());
+}
+
TEST_F(StatsCounterTest, VerifyProcessInterval) {
StatsCounterObserverImpl* observer = new StatsCounterObserverImpl();
AvgCounter counter(&clock_, observer);
