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Add support for updating histogram for received fraction loss ("WebRTC.Video.ReceivedPacketsLostInPercent") when running new video api.

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Add tests for verifying that video histograms are updated.

BUG=
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/44309004

Cr-Commit-Position: refs/heads/master@{#9085}
This commit is contained in:
Åsa Persson 2015-04-27 10:09:49 +02:00
parent 52ef9d7738
commit 3c391cbabb
3 changed files with 183 additions and 3 deletions
Show Stats Download Patch File Download Diff File Expand all files Collapse all files

165
webrtc/video/end_to_end_tests.cc
View File

@ -78,6 +78,7 @@ class EndToEndTest : public test::CallTest {
void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
void TestRtpStatePreservation(bool use_rtx);
void TestReceivedFecPacketsNotNacked(const FakeNetworkPipe::Config& config);
void VerifyHistogramStats(bool use_rtx, bool use_red);
};
TEST_F(EndToEndTest, ReceiverCanBeStartedTwice) {
@ -1497,11 +1498,12 @@ TEST_F(EndToEndTest, VerifyNackStats) {
int64_t start_runtime_ms_;
} test;
test::ClearHistograms();
RunBaseTest(&test);
EXPECT_NE(-1, test::LastHistogramSample(
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.UniqueNackRequestsSentInPercent"));
EXPECT_NE(-1, test::LastHistogramSample(
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
EXPECT_GT(test::LastHistogramSample(
"WebRTC.Video.NackPacketsSentPerMinute"), 0);
@ -1509,6 +1511,165 @@ TEST_F(EndToEndTest, VerifyNackStats) {
"WebRTC.Video.NackPacketsReceivedPerMinute"), 0);
}
void EndToEndTest::VerifyHistogramStats(bool use_rtx, bool use_red) {
class StatsObserver : public test::EndToEndTest, public PacketReceiver {
public:
StatsObserver(bool use_rtx, bool use_red)
: EndToEndTest(kLongTimeoutMs),
use_rtx_(use_rtx),
use_red_(use_red),
sender_call_(nullptr),
receiver_call_(nullptr),
start_runtime_ms_(-1) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (MinMetricRunTimePassed())
observation_complete_->Set();
return SEND_PACKET;
}
DeliveryStatus DeliverPacket(const uint8_t* packet,
size_t length) override {
// GetStats calls GetSendChannelRtcpStatistics
// (via VideoSendStream::GetRtt) which updates ReportBlockStats used by
// WebRTC.Video.SentPacketsLostInPercent.
// TODO(asapersson): Remove dependency on calling GetStats.
sender_call_->GetStats();
return receiver_call_->Receiver()->DeliverPacket(packet, length);
}
bool MinMetricRunTimePassed() {
int64_t now = Clock::GetRealTimeClock()->TimeInMilliseconds();
if (start_runtime_ms_ == -1) {
start_runtime_ms_ = now;
return false;
}
int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
return elapsed_sec > metrics::kMinRunTimeInSeconds * 2;
}
void ModifyConfigs(VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// NACK
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
// FEC
if (use_red_) {
send_config->rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.fec.ulpfec_payload_type = kUlpfecPayloadType;
}
// RTX
if (use_rtx_) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].ssrc =
kSendRtxSsrcs[0];
(*receive_configs)[0].rtp.rtx[kFakeSendPayloadType].payload_type =
kSendRtxPayloadType;
}
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void SetReceivers(PacketReceiver* send_transport_receiver,
PacketReceiver* receive_transport_receiver) override {
test::RtpRtcpObserver::SetReceivers(this, receive_transport_receiver);
}
void PerformTest() override {
EXPECT_EQ(kEventSignaled, Wait())
<< "Timed out waiting for packet to be NACKed.";
}
bool use_rtx_;
bool use_red_;
Call* sender_call_;
Call* receiver_call_;
int64_t start_runtime_ms_;
} test(use_rtx, use_red);
test::ClearHistograms();
RunBaseTest(&test);
// Verify that stats have been updated once.
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.NackPacketsReceivedPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.FirPacketsReceivedPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PliPacketsSentPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PliPacketsReceivedPerMinute"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.KeyFramesSentInPermille"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.KeyFramesReceivedInPermille"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.SentPacketsLostInPercent"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.ReceivedPacketsLostInPercent"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.BitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.BitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.MediaBitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.MediaBitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PaddingBitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.PaddingBitrateReceivedInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.RetransmittedBitrateSentInKbps"));
EXPECT_EQ(1, test::NumHistogramSamples(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
int num_rtx_samples = use_rtx ? 1 : 0;
EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
"WebRTC.Video.RtxBitrateSentInKbps"));
EXPECT_EQ(num_rtx_samples, test::NumHistogramSamples(
"WebRTC.Video.RtxBitrateReceivedInKbps"));
int num_red_samples = use_red ? 1 : 0;
EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
"WebRTC.Video.FecBitrateSentInKbps"));
EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
"WebRTC.Video.FecBitrateReceivedInKbps"));
EXPECT_EQ(num_red_samples, test::NumHistogramSamples(
"WebRTC.Video.ReceivedFecPacketsInPercent"));
}
TEST_F(EndToEndTest, VerifyHistogramStatsWithRtx) {
const bool kEnabledRtx = true;
const bool kEnabledRed = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed);
}
TEST_F(EndToEndTest, VerifyHistogramStatsWithRed) {
const bool kEnabledRtx = false;
const bool kEnabledRed = true;
VerifyHistogramStats(kEnabledRtx, kEnabledRed);
}
void EndToEndTest::TestXrReceiverReferenceTimeReport(bool enable_rrtr) {
static const int kNumRtcpReportPacketsToObserve = 5;
class RtcpXrObserver : public test::EndToEndTest {

18
webrtc/video/receive_statistics_proxy.cc
View File

@ -12,6 +12,7 @@
#include "webrtc/system_wrappers/interface/clock.h"
#include "webrtc/system_wrappers/interface/critical_section_wrapper.h"
#include "webrtc/system_wrappers/interface/metrics.h"
namespace webrtc {
@ -24,7 +25,21 @@ ReceiveStatisticsProxy::ReceiveStatisticsProxy(uint32_t ssrc, Clock* clock)
stats_.ssrc = ssrc;
}
ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {}
ReceiveStatisticsProxy::~ReceiveStatisticsProxy() {
UpdateHistograms();
}
void ReceiveStatisticsProxy::UpdateHistograms() const {
int fraction_lost;
{
CriticalSectionScoped lock(crit_.get());
fraction_lost = report_block_stats_.FractionLostInPercent();
}
if (fraction_lost != -1) {
RTC_HISTOGRAM_PERCENTAGE("WebRTC.Video.ReceivedPacketsLostInPercent",
fraction_lost);
}
}
VideoReceiveStream::Stats ReceiveStatisticsProxy::GetStats() const {
CriticalSectionScoped lock(crit_.get());
@ -74,6 +89,7 @@ void ReceiveStatisticsProxy::StatisticsUpdated(
if (stats_.ssrc != ssrc)
return;
stats_.rtcp_stats = statistics;
report_block_stats_.Store(statistics, ssrc, 0);
}
void ReceiveStatisticsProxy::CNameChanged(const char* cname, uint32_t ssrc) {

3
webrtc/video/receive_statistics_proxy.h
View File

@ -20,6 +20,7 @@
#include "webrtc/modules/video_coding/main/interface/video_coding_defines.h"
#include "webrtc/video_engine/include/vie_codec.h"
#include "webrtc/video_engine/include/vie_rtp_rtcp.h"
#include "webrtc/video_engine/report_block_stats.h"
#include "webrtc/video_receive_stream.h"
#include "webrtc/video_renderer.h"
@ -78,12 +79,14 @@ class ReceiveStatisticsProxy : public ViEDecoderObserver,
uint32_t ssrc) override;
private:
void UpdateHistograms() const;
Clock* const clock_;
rtc::scoped_ptr<CriticalSectionWrapper> crit_;
VideoReceiveStream::Stats stats_ GUARDED_BY(crit_);
RateStatistics decode_fps_estimator_ GUARDED_BY(crit_);
RateStatistics renders_fps_estimator_ GUARDED_BY(crit_);
ReportBlockStats report_block_stats_ GUARDED_BY(crit_);
};
} // namespace webrtc