Add stats for video:
- number of sent/received RTCP NACK/FIR/PLI per minute
- percentage of unique sent/received NACK requests
- percentage of discarded/duplicated packets by the jitter buffer
- permille of sent/received key frames
BUG=crbug/419657
R=mflodman@webrtc.org, stefan@webrtc.org
Review URL: https://webrtc-codereview.appspot.com/24959004
git-svn-id: http://webrtc.googlecode.com/svn/trunk@7592 4adac7df-926f-26a2-2b94-8c16560cd09d
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.cc b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
index d09fccd..e3a4a8a 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.cc
@@ -25,6 +25,7 @@
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/event_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
namespace webrtc {
@@ -143,6 +144,8 @@
drop_count_(0),
num_consecutive_old_frames_(0),
num_consecutive_old_packets_(0),
+ num_packets_(0),
+ num_duplicated_packets_(0),
num_discarded_packets_(0),
jitter_estimate_(clock),
inter_frame_delay_(clock_->TimeInMilliseconds()),
@@ -190,6 +193,8 @@
drop_count_ = rhs.drop_count_;
num_consecutive_old_frames_ = rhs.num_consecutive_old_frames_;
num_consecutive_old_packets_ = rhs.num_consecutive_old_packets_;
+ num_packets_ = rhs.num_packets_;
+ num_duplicated_packets_ = rhs.num_duplicated_packets_;
num_discarded_packets_ = rhs.num_discarded_packets_;
jitter_estimate_ = rhs.jitter_estimate_;
inter_frame_delay_ = rhs.inter_frame_delay_;
@@ -238,6 +243,15 @@
}
}
+void VCMJitterBuffer::UpdateHistograms() {
+ if (num_packets_ > 0) {
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DiscardedPacketsInPercent",
+ num_discarded_packets_ * 100 / num_packets_);
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.DuplicatedPacketsInPercent",
+ num_duplicated_packets_ * 100 / num_packets_);
+ }
+}
+
void VCMJitterBuffer::Start() {
CriticalSectionScoped cs(crit_sect_);
running_ = true;
@@ -250,6 +264,8 @@
num_consecutive_old_frames_ = 0;
num_consecutive_old_packets_ = 0;
+ num_packets_ = 0;
+ num_duplicated_packets_ = 0;
num_discarded_packets_ = 0;
// Start in a non-signaled state.
@@ -265,6 +281,7 @@
void VCMJitterBuffer::Stop() {
crit_sect_->Enter();
+ UpdateHistograms();
running_ = false;
last_decoded_state_.Reset();
free_frames_.clear();
@@ -313,6 +330,16 @@
return receive_statistics_;
}
+int VCMJitterBuffer::num_packets() const {
+ CriticalSectionScoped cs(crit_sect_);
+ return num_packets_;
+}
+
+int VCMJitterBuffer::num_duplicated_packets() const {
+ CriticalSectionScoped cs(crit_sect_);
+ return num_duplicated_packets_;
+}
+
int VCMJitterBuffer::num_discarded_packets() const {
CriticalSectionScoped cs(crit_sect_);
return num_discarded_packets_;
@@ -543,6 +570,7 @@
// Gets frame to use for this timestamp. If no match, get empty frame.
VCMFrameBufferEnum VCMJitterBuffer::GetFrame(const VCMPacket& packet,
VCMFrameBuffer** frame) {
+ ++num_packets_;
// Does this packet belong to an old frame?
if (last_decoded_state_.IsOldPacket(&packet)) {
// Account only for media packets.
@@ -747,6 +775,7 @@
case kNoError:
case kOutOfBoundsPacket:
case kDuplicatePacket: {
+ ++num_duplicated_packets_;
break;
}
case kFlushIndicator:
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer.h b/webrtc/modules/video_coding/main/source/jitter_buffer.h
index 6ed9cfb..182b80b 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer.h
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer.h
@@ -103,6 +103,12 @@
// won't be able to decode them.
int num_not_decodable_packets() const;
+ // Gets number of packets received.
+ int num_packets() const;
+
+ // Gets number of duplicated packets received.
+ int num_duplicated_packets() const;
+
// Gets number of packets discarded by the jitter buffer.
int num_discarded_packets() const;
@@ -271,6 +277,8 @@
uint16_t EstimatedLowSequenceNumber(const VCMFrameBuffer& frame) const;
+ void UpdateHistograms();
+
Clock* clock_;
// If we are running (have started) or not.
bool running_;
@@ -303,6 +311,10 @@
int num_consecutive_old_frames_;
// Number of packets in a row that have been too old.
int num_consecutive_old_packets_;
+ // Number of packets received.
+ int num_packets_;
+ // Number of duplicated packets received.
+ int num_duplicated_packets_;
// Number of packets discarded by the jitter buffer.
int num_discarded_packets_;
diff --git a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
index 0490658..899a3ee 100644
--- a/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
+++ b/webrtc/modules/video_coding/main/source/jitter_buffer_unittest.cc
@@ -512,6 +512,8 @@
packet_->markerBit = false;
packet_->seqNum = seq_num_;
packet_->timestamp = timestamp_;
+ EXPECT_EQ(0, jitter_buffer_->num_packets());
+ EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
bool retransmitted = false;
EXPECT_EQ(kIncomplete, jitter_buffer_->InsertPacket(*packet_,
@@ -520,6 +522,8 @@
VCMEncodedFrame* frame_out = DecodeCompleteFrame();
EXPECT_TRUE(frame_out == NULL);
+ EXPECT_EQ(1, jitter_buffer_->num_packets());
+ EXPECT_EQ(0, jitter_buffer_->num_duplicated_packets());
packet_->isFirstPacket = false;
packet_->markerBit = true;
@@ -527,6 +531,8 @@
// Insert a packet into a frame.
EXPECT_EQ(kDuplicatePacket, jitter_buffer_->InsertPacket(*packet_,
&retransmitted));
+ EXPECT_EQ(2, jitter_buffer_->num_packets());
+ EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
seq_num_++;
packet_->seqNum = seq_num_;
@@ -539,6 +545,8 @@
CheckOutFrame(frame_out, 2 * size_, false);
EXPECT_EQ(kVideoFrameKey, frame_out->FrameType());
+ EXPECT_EQ(3, jitter_buffer_->num_packets());
+ EXPECT_EQ(1, jitter_buffer_->num_duplicated_packets());
}
TEST_F(TestBasicJitterBuffer, H264InsertStartCode) {
diff --git a/webrtc/system_wrappers/interface/metrics.h b/webrtc/system_wrappers/interface/metrics.h
index be9564a..0390140 100644
--- a/webrtc/system_wrappers/interface/metrics.h
+++ b/webrtc/system_wrappers/interface/metrics.h
@@ -69,12 +69,26 @@
// Also consider changing string to const char* when switching to atomics.
// Histogram for counters.
+#define RTC_HISTOGRAM_COUNTS_100(name, sample) RTC_HISTOGRAM_COUNTS( \
+ name, sample, 1, 100, 50)
+
+#define RTC_HISTOGRAM_COUNTS_1000(name, sample) RTC_HISTOGRAM_COUNTS( \
+ name, sample, 1, 1000, 50)
+
+#define RTC_HISTOGRAM_COUNTS_10000(name, sample) RTC_HISTOGRAM_COUNTS( \
+ name, sample, 1, 10000, 50)
+
#define RTC_HISTOGRAM_COUNTS(name, sample, min, max, bucket_count) \
RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
webrtc::metrics::HistogramFactoryGetCounts( \
name, min, max, bucket_count))
+// Histogram for percentage.
+#define RTC_HISTOGRAM_PERCENTAGE(name, sample) \
+ RTC_HISTOGRAM_ENUMERATION(name, sample, 101)
+
// Histogram for enumerators.
+// |boundary| should be above the max enumerator sample.
#define RTC_HISTOGRAM_ENUMERATION(name, sample, boundary) \
RTC_HISTOGRAM_COMMON_BLOCK(name, sample, \
webrtc::metrics::HistogramFactoryGetEnumeration(name, boundary))
diff --git a/webrtc/video_engine/vie_channel.cc b/webrtc/video_engine/vie_channel.cc
index bbcb602..ded2140 100644
--- a/webrtc/video_engine/vie_channel.cc
+++ b/webrtc/video_engine/vie_channel.cc
@@ -25,6 +25,7 @@
#include "webrtc/modules/video_render/include/video_render_defines.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
#include "webrtc/system_wrappers/interface/thread_wrapper.h"
#include "webrtc/video_engine/call_stats.h"
#include "webrtc/video_engine/include/vie_codec.h"
@@ -147,7 +148,8 @@
sender_(sender),
nack_history_size_sender_(kSendSidePacketHistorySize),
max_nack_reordering_threshold_(kMaxPacketAgeToNack),
- pre_render_callback_(NULL) {
+ pre_render_callback_(NULL),
+ start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
RtpRtcp::Configuration configuration;
configuration.id = ViEModuleId(engine_id, channel_id);
configuration.audio = false;
@@ -222,6 +224,7 @@
}
ViEChannel::~ViEChannel() {
+ UpdateHistograms();
// Make sure we don't get more callbacks from the RTP module.
module_process_thread_.DeRegisterModule(vie_receiver_.GetReceiveStatistics());
module_process_thread_.DeRegisterModule(rtp_rtcp_.get());
@@ -246,6 +249,55 @@
VideoCodingModule::Destroy(vcm_);
}
+void ViEChannel::UpdateHistograms() {
+ const float kMinCallLengthInMinutes = 0.5f;
+ float elapsed_minutes =
+ (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
+ if (elapsed_minutes < kMinCallLengthInMinutes) {
+ return;
+ }
+ RtcpPacketTypeCounter rtcp_sent;
+ RtcpPacketTypeCounter rtcp_received;
+ GetRtcpPacketTypeCounters(&rtcp_sent, &rtcp_received);
+
+ if (sender_) {
+ if (rtcp_received.nack_requests > 0) {
+ RTC_HISTOGRAM_PERCENTAGE(
+ "WebRTC.Video.UniqueNackRequestsReceivedInPercent",
+ rtcp_received.UniqueNackRequestsInPercent());
+ }
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsReceivedPerMinute",
+ rtcp_received.nack_packets / elapsed_minutes);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsReceivedPerMinute",
+ rtcp_received.fir_packets / elapsed_minutes);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsReceivedPerMinute",
+ rtcp_received.pli_packets / elapsed_minutes);
+ } else if (vie_receiver_.GetRemoteSsrc() > 0) {
+ // Get receive stats if we are receiving packets, i.e. there is a remote
+ // ssrc.
+ if (rtcp_sent.nack_requests > 0) {
+ RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.UniqueNackRequestsSentInPercent",
+ rtcp_sent.UniqueNackRequestsInPercent());
+ }
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.NackPacketsSentPerMinute",
+ rtcp_sent.nack_packets / elapsed_minutes);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.FirPacketsSentPerMinute",
+ rtcp_sent.fir_packets / elapsed_minutes);
+ RTC_HISTOGRAM_COUNTS_10000("WebRTC.Video.PliPacketsSentPerMinute",
+ rtcp_sent.pli_packets / elapsed_minutes);
+
+ webrtc::VCMFrameCount frames;
+ if (vcm_->ReceivedFrameCount(frames) == VCM_OK) {
+ uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
+ if (total_frames > 0) {
+ RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesReceivedInPermille",
+ static_cast<int>((frames.numKeyFrames * 1000.0f / total_frames) +
+ 0.5f));
+ }
+ }
+ }
+}
+
int32_t ViEChannel::SetSendCodec(const VideoCodec& video_codec,
bool new_stream) {
if (!sender_) {
diff --git a/webrtc/video_engine/vie_channel.h b/webrtc/video_engine/vie_channel.h
index e55ade7..0327906 100644
--- a/webrtc/video_engine/vie_channel.h
+++ b/webrtc/video_engine/vie_channel.h
@@ -383,6 +383,8 @@
int GetRequiredNackListSize(int target_delay_ms);
void SetRtxSendStatus(bool enable);
+ void UpdateHistograms();
+
// ViEChannel exposes methods that allow to modify observers and callbacks
// to be modified. Such an API-style is cumbersome to implement and maintain
// at all the levels when comparing to only setting them at construction. As
@@ -499,6 +501,7 @@
int nack_history_size_sender_;
int max_nack_reordering_threshold_;
I420FrameCallback* pre_render_callback_;
+ const int64_t start_ms_;
std::map<uint32_t, RTCPReportBlock> prev_report_blocks_;
};
diff --git a/webrtc/video_engine/vie_encoder.cc b/webrtc/video_engine/vie_encoder.cc
index 26ad8ac..9d6da97 100644
--- a/webrtc/video_engine/vie_encoder.cc
+++ b/webrtc/video_engine/vie_encoder.cc
@@ -26,6 +26,7 @@
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/logging.h"
+#include "webrtc/system_wrappers/interface/metrics.h"
#include "webrtc/system_wrappers/interface/tick_util.h"
#include "webrtc/system_wrappers/interface/trace_event.h"
#include "webrtc/video_engine/include/vie_codec.h"
@@ -156,7 +157,8 @@
picture_id_rpsi_(0),
qm_callback_(NULL),
video_suspended_(false),
- pre_encode_callback_(NULL) {
+ pre_encode_callback_(NULL),
+ start_ms_(Clock::GetRealTimeClock()->TimeInMilliseconds()) {
RtpRtcp::Configuration configuration;
configuration.id = ViEModuleId(engine_id_, channel_id_);
configuration.audio = false; // Video.
@@ -225,6 +227,7 @@
}
ViEEncoder::~ViEEncoder() {
+ UpdateHistograms();
if (bitrate_controller_) {
bitrate_controller_->RemoveBitrateObserver(bitrate_observer_.get());
}
@@ -237,6 +240,25 @@
delete qm_callback_;
}
+void ViEEncoder::UpdateHistograms() {
+ const float kMinCallLengthInMinutes = 0.5f;
+ float elapsed_minutes =
+ (Clock::GetRealTimeClock()->TimeInMilliseconds() - start_ms_) / 60000.0f;
+ if (elapsed_minutes < kMinCallLengthInMinutes) {
+ return;
+ }
+ webrtc::VCMFrameCount frames;
+ if (vcm_.SentFrameCount(frames) != VCM_OK) {
+ return;
+ }
+ uint32_t total_frames = frames.numKeyFrames + frames.numDeltaFrames;
+ if (total_frames > 0) {
+ RTC_HISTOGRAM_COUNTS_1000("WebRTC.Video.KeyFramesSentInPermille",
+ static_cast<int>(
+ (frames.numKeyFrames * 1000.0f / total_frames) + 0.5f));
+ }
+}
+
int ViEEncoder::Owner() const {
return channel_id_;
}
diff --git a/webrtc/video_engine/vie_encoder.h b/webrtc/video_engine/vie_encoder.h
index 51c1d01..36f87fa 100644
--- a/webrtc/video_engine/vie_encoder.h
+++ b/webrtc/video_engine/vie_encoder.h
@@ -193,6 +193,8 @@
private:
bool EncoderPaused() const EXCLUSIVE_LOCKS_REQUIRED(data_cs_);
+ void UpdateHistograms();
+
int32_t engine_id_;
const int channel_id_;
const uint32_t number_of_cores_;
@@ -235,6 +237,7 @@
QMVideoSettingsCallback* qm_callback_;
bool video_suspended_ GUARDED_BY(data_cs_);
I420FrameCallback* pre_encode_callback_ GUARDED_BY(callback_cs_);
+ const int64_t start_ms_;
};
} // namespace webrtc
