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webrtc/video/end_to_end_tests/stats_tests.cc
Per K 217b384c1b Remove rtp header extension from config of Call audio and video receivers 
These configurations are no longer used by call. Header extensions are identified once when demuxing packets in WebrtcVideoEngine::OnPacketReceived and WebrtcVoiceEngine::OnPacketReceived.

Change-Id: I49de9005f0aa9ab32f2c5d3abcdd8bd12343022d
Bug: webrtc:7135
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291480
Owners-Override: Per Kjellander <perkj@webrtc.org>
Commit-Queue: Per Kjellander <perkj@webrtc.org>
Reviewed-by: Erik Språng <sprang@webrtc.org>
Reviewed-by: Henrik Boström <hbos@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39236}
2023-01-31 11:58:43 +00:00

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/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include "absl/algorithm/container.h"
#include "absl/types/optional.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "call/fake_network_pipe.h"
#include "call/simulated_network.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "rtc_base/strings/string_builder.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/fake_encoder.h"
#include "test/gtest.h"
#include "test/rtcp_packet_parser.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kVideoContentTypeExtensionId = 1,
};
} // namespace
class StatsEndToEndTest : public test::CallTest {
public:
StatsEndToEndTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kVideoContentTypeUri,
kVideoContentTypeExtensionId));
}
};
TEST_F(StatsEndToEndTest, GetStats) {
static const int kStartBitrateBps = 3000000;
static const int kExpectedRenderDelayMs = 20;
class StatsObserver : public test::EndToEndTest {
public:
StatsObserver()
: EndToEndTest(kLongTimeout), encoder_factory_([]() {
return std::make_unique<test::DelayedEncoder>(
Clock::GetRealTimeClock(), 10);
}) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
// Drop every 25th packet => 4% loss.
static const int kPacketLossFrac = 25;
RtpPacket header;
if (header.Parse(packet, length) &&
expected_send_ssrcs_.find(header.Ssrc()) !=
expected_send_ssrcs_.end() &&
header.SequenceNumber() % kPacketLossFrac == 0) {
return DROP_PACKET;
}
check_stats_event_.Set();
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
check_stats_event_.Set();
return SEND_PACKET;
}
Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
check_stats_event_.Set();
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
check_stats_event_.Set();
return SEND_PACKET;
}
bool CheckReceiveStats() {
for (size_t i = 0; i < receive_streams_.size(); ++i) {
VideoReceiveStreamInterface::Stats stats =
receive_streams_[i]->GetStats();
EXPECT_EQ(expected_receive_ssrcs_[i], stats.ssrc);
// Make sure all fields have been populated.
// TODO(pbos): Use CompoundKey if/when we ever know that all stats are
// always filled for all receivers.
receive_stats_filled_["IncomingRate"] |=
stats.network_frame_rate != 0 || stats.total_bitrate_bps != 0;
send_stats_filled_["DecoderImplementationName"] |=
stats.decoder_implementation_name ==
test::FakeDecoder::kImplementationName;
receive_stats_filled_["PowerEfficientDecoder"] =
stats.power_efficient_decoder.has_value();
receive_stats_filled_["RenderDelayAsHighAsExpected"] |=
stats.render_delay_ms >= kExpectedRenderDelayMs;
receive_stats_filled_["FrameCallback"] |= stats.decode_frame_rate != 0;
receive_stats_filled_["FrameRendered"] |= stats.render_frame_rate != 0;
receive_stats_filled_["StatisticsUpdated"] |=
stats.rtp_stats.packets_lost != 0 || stats.rtp_stats.jitter != 0;
receive_stats_filled_["DataCountersUpdated"] |=
stats.rtp_stats.packet_counter.payload_bytes != 0 ||
stats.rtp_stats.packet_counter.header_bytes != 0 ||
stats.rtp_stats.packet_counter.packets != 0 ||
stats.rtp_stats.packet_counter.padding_bytes != 0;
receive_stats_filled_["CodecStats"] |= stats.target_delay_ms != 0;
receive_stats_filled_["FrameCounts"] |=
stats.frame_counts.key_frames != 0 ||
stats.frame_counts.delta_frames != 0;
receive_stats_filled_["CName"] |= !stats.c_name.empty();
receive_stats_filled_["RtcpPacketTypeCount"] |=
stats.rtcp_packet_type_counts.fir_packets != 0 ||
stats.rtcp_packet_type_counts.nack_packets != 0 ||
stats.rtcp_packet_type_counts.pli_packets != 0 ||
stats.rtcp_packet_type_counts.nack_requests != 0 ||
stats.rtcp_packet_type_counts.unique_nack_requests != 0;
RTC_DCHECK(stats.current_payload_type == -1 ||
stats.current_payload_type == kFakeVideoSendPayloadType);
receive_stats_filled_["IncomingPayloadType"] |=
stats.current_payload_type == kFakeVideoSendPayloadType;
}
return AllStatsFilled(receive_stats_filled_);
}
bool CheckSendStats() {
RTC_DCHECK(send_stream_);
VideoSendStream::Stats stats;
SendTask(task_queue_, [&]() { stats = send_stream_->GetStats(); });
size_t expected_num_streams =
kNumSimulcastStreams + expected_send_ssrcs_.size();
send_stats_filled_["NumStreams"] |=
stats.substreams.size() == expected_num_streams;
send_stats_filled_["CpuOveruseMetrics"] |=
stats.avg_encode_time_ms != 0 && stats.encode_usage_percent != 0 &&
stats.total_encode_time_ms != 0;
send_stats_filled_["EncoderImplementationName"] |=
stats.encoder_implementation_name ==
test::FakeEncoder::kImplementationName;
send_stats_filled_["PowerEfficientEncoder"] |=
stats.power_efficient_encoder == true;
for (const auto& kv : stats.substreams) {
if (expected_send_ssrcs_.find(kv.first) == expected_send_ssrcs_.end())
continue; // Probably RTX.
send_stats_filled_[CompoundKey("CapturedFrameRate", kv.first)] |=
stats.input_frame_rate != 0;
const VideoSendStream::StreamStats& stream_stats = kv.second;
send_stats_filled_[CompoundKey("StatisticsUpdated", kv.first)] |=
stream_stats.report_block_data.has_value();
send_stats_filled_[CompoundKey("DataCountersUpdated", kv.first)] |=
stream_stats.rtp_stats.fec.packets != 0 ||
stream_stats.rtp_stats.transmitted.padding_bytes != 0 ||
stream_stats.rtp_stats.retransmitted.packets != 0 ||
stream_stats.rtp_stats.transmitted.packets != 0;
send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Total",
kv.first)] |=
stream_stats.total_bitrate_bps != 0;
send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Retransmit",
kv.first)] |=
stream_stats.retransmit_bitrate_bps != 0;
send_stats_filled_[CompoundKey("FrameCountObserver", kv.first)] |=
stream_stats.frame_counts.delta_frames != 0 ||
stream_stats.frame_counts.key_frames != 0;
send_stats_filled_[CompoundKey("OutgoingRate", kv.first)] |=
stats.encode_frame_rate != 0;
send_stats_filled_[CompoundKey("Delay", kv.first)] |=
stream_stats.avg_delay_ms != 0 || stream_stats.max_delay_ms != 0;
// TODO(pbos): Use CompoundKey when the test makes sure that all SSRCs
// report dropped packets.
send_stats_filled_["RtcpPacketTypeCount"] |=
stream_stats.rtcp_packet_type_counts.fir_packets != 0 ||
stream_stats.rtcp_packet_type_counts.nack_packets != 0 ||
stream_stats.rtcp_packet_type_counts.pli_packets != 0 ||
stream_stats.rtcp_packet_type_counts.nack_requests != 0 ||
stream_stats.rtcp_packet_type_counts.unique_nack_requests != 0;
}
return AllStatsFilled(send_stats_filled_);
}
std::string CompoundKey(const char* name, uint32_t ssrc) {
rtc::StringBuilder oss;
oss << name << "_" << ssrc;
return oss.Release();
}
bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
for (const auto& stat : stats_map) {
if (!stat.second)
return false;
}
return true;
}
BuiltInNetworkBehaviorConfig GetSendTransportConfig() const override {
BuiltInNetworkBehaviorConfig network_config;
network_config.loss_percent = 5;
return network_config;
}
void ModifySenderBitrateConfig(
BitrateConstraints* bitrate_config) override {
bitrate_config->start_bitrate_bps = kStartBitrateBps;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
encoder_config->max_bitrate_bps = 50000;
for (auto& layer : encoder_config->simulcast_layers) {
layer.min_bitrate_bps = 10000;
layer.target_bitrate_bps = 15000;
layer.max_bitrate_bps = 20000;
}
send_config->rtp.c_name = "SomeCName";
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
const std::vector<uint32_t>& ssrcs = send_config->rtp.ssrcs;
for (size_t i = 0; i < ssrcs.size(); ++i) {
expected_send_ssrcs_.insert(ssrcs[i]);
expected_receive_ssrcs_.push_back(
(*receive_configs)[i].rtp.remote_ssrc);
(*receive_configs)[i].render_delay_ms = kExpectedRenderDelayMs;
(*receive_configs)[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[i].rtp.rtx_ssrc = kSendRtxSsrcs[i];
(*receive_configs)[i]
.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
kFakeVideoSendPayloadType;
}
for (size_t i = 0; i < kNumSimulcastStreams; ++i)
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
// Use a delayed encoder to make sure we see CpuOveruseMetrics stats that
// are non-zero.
send_config->encoder_settings.encoder_factory = &encoder_factory_;
}
size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override {
send_stream_ = send_stream;
receive_streams_ = receive_streams;
task_queue_ = TaskQueueBase::Current();
}
void PerformTest() override {
Clock* clock = Clock::GetRealTimeClock();
int64_t now_ms = clock->TimeInMilliseconds();
int64_t stop_time_ms = now_ms + test::CallTest::kLongTimeout.ms();
bool receive_ok = false;
bool send_ok = false;
while (now_ms < stop_time_ms) {
if (!receive_ok && task_queue_) {
SendTask(task_queue_, [&]() { receive_ok = CheckReceiveStats(); });
}
if (!send_ok)
send_ok = CheckSendStats();
if (receive_ok && send_ok)
return;
int64_t time_until_timeout_ms = stop_time_ms - now_ms;
if (time_until_timeout_ms > 0)
check_stats_event_.Wait(TimeDelta::Millis(time_until_timeout_ms));
now_ms = clock->TimeInMilliseconds();
}
ADD_FAILURE() << "Timed out waiting for filled stats.";
for (const auto& kv : receive_stats_filled_) {
if (!kv.second) {
ADD_FAILURE() << "Missing receive stats: " << kv.first;
}
}
for (const auto& kv : send_stats_filled_) {
if (!kv.second) {
ADD_FAILURE() << "Missing send stats: " << kv.first;
}
}
}
test::FunctionVideoEncoderFactory encoder_factory_;
std::vector<VideoReceiveStreamInterface*> receive_streams_;
std::map<std::string, bool> receive_stats_filled_;
VideoSendStream* send_stream_ = nullptr;
std::map<std::string, bool> send_stats_filled_;
std::vector<uint32_t> expected_receive_ssrcs_;
std::set<uint32_t> expected_send_ssrcs_;
rtc::Event check_stats_event_;
TaskQueueBase* task_queue_ = nullptr;
} test;
RunBaseTest(&test);
}
TEST_F(StatsEndToEndTest, TimingFramesAreReported) {
static const int kExtensionId = 5;
RegisterRtpExtension(
RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
class StatsObserver : public test::EndToEndTest {
public:
StatsObserver() : EndToEndTest(kLongTimeout) {}
private:
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
}
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override {
receive_streams_ = receive_streams;
task_queue_ = TaskQueueBase::Current();
}
void PerformTest() override {
// No frames reported initially.
SendTask(task_queue_, [&]() {
for (const auto& receive_stream : receive_streams_) {
EXPECT_FALSE(receive_stream->GetStats().timing_frame_info);
}
});
// Wait for at least one timing frame to be sent with 100ms grace period.
SleepMs(kDefaultTimingFramesDelayMs + 100);
// Check that timing frames are reported for each stream.
SendTask(task_queue_, [&]() {
for (const auto& receive_stream : receive_streams_) {
EXPECT_TRUE(receive_stream->GetStats().timing_frame_info);
}
});
}
std::vector<VideoReceiveStreamInterface*> receive_streams_;
TaskQueueBase* task_queue_ = nullptr;
} test;
RunBaseTest(&test);
}
TEST_F(StatsEndToEndTest, TestReceivedRtpPacketStats) {
static const size_t kNumRtpPacketsToSend = 5;
class ReceivedRtpStatsObserver : public test::EndToEndTest {
public:
explicit ReceivedRtpStatsObserver(TaskQueueBase* task_queue)
: EndToEndTest(kDefaultTimeout), task_queue_(task_queue) {}
private:
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override {
receive_stream_ = receive_streams[0];
}
void OnStreamsStopped() override { task_safety_flag_->SetNotAlive(); }
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (sent_rtp_ >= kNumRtpPacketsToSend) {
// Need to check the stats on the correct thread.
task_queue_->PostTask(SafeTask(task_safety_flag_, [this]() {
VideoReceiveStreamInterface::Stats stats =
receive_stream_->GetStats();
if (kNumRtpPacketsToSend == stats.rtp_stats.packet_counter.packets) {
observation_complete_.Set();
}
}));
return DROP_PACKET;
}
++sent_rtp_;
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while verifying number of received RTP packets.";
}
VideoReceiveStreamInterface* receive_stream_ = nullptr;
uint32_t sent_rtp_ = 0;
TaskQueueBase* const task_queue_;
rtc::scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_ =
PendingTaskSafetyFlag::CreateDetached();
} test(task_queue());
RunBaseTest(&test);
}
#if defined(WEBRTC_WIN)
// Disabled due to flakiness on Windows (bugs.webrtc.org/7483).
#define MAYBE_ContentTypeSwitches DISABLED_ContentTypeSwitches
#else
#define MAYBE_ContentTypeSwitches ContentTypeSwitches
#endif
TEST_F(StatsEndToEndTest, MAYBE_ContentTypeSwitches) {
class StatsObserver : public test::BaseTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver() : BaseTest(kLongTimeout), num_frames_received_(0) {}
bool ShouldCreateReceivers() const override { return true; }
void OnFrame(const VideoFrame& video_frame) override {
// The RTT is needed to estimate `ntp_time_ms` which is used by
// end-to-end delay stats. Therefore, start counting received frames once
// `ntp_time_ms` is valid.
if (video_frame.ntp_time_ms() > 0 &&
Clock::GetRealTimeClock()->CurrentNtpInMilliseconds() >=
video_frame.ntp_time_ms()) {
MutexLock lock(&mutex_);
++num_frames_received_;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (MinNumberOfFramesReceived())
observation_complete_.Set();
return SEND_PACKET;
}
bool MinNumberOfFramesReceived() const {
// Have some room for frames with wrong content type during switch.
const int kMinRequiredHistogramSamples = 200 + 50;
MutexLock lock(&mutex_);
return num_frames_received_ > kMinRequiredHistogramSamples;
}
// May be called several times.
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for enough packets.";
// Reset frame counter so next PerformTest() call will do something.
{
MutexLock lock(&mutex_);
num_frames_received_ = 0;
}
}
mutable Mutex mutex_;
int num_frames_received_ RTC_GUARDED_BY(&mutex_);
} test;
metrics::Reset();
Call::Config send_config(send_event_log_.get());
test.ModifySenderBitrateConfig(&send_config.bitrate_config);
Call::Config recv_config(recv_event_log_.get());
test.ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
VideoEncoderConfig encoder_config_with_screenshare;
SendTask(
task_queue(), [this, &test, &send_config, &recv_config,
&encoder_config_with_screenshare]() {
CreateSenderCall(send_config);
CreateReceiverCall(recv_config);
CreateReceiveTransport(test.GetReceiveTransportConfig(), &test);
CreateSendTransport(test.GetReceiveTransportConfig(), &test);
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
CreateSendConfig(1, 0, 0);
CreateMatchingReceiveConfigs();
// Modify send and receive configs.
GetVideoSendConfig()->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[0].renderer = &test;
// RTT needed for RemoteNtpTimeEstimator for the receive stream.
video_receive_configs_[0].rtp.rtcp_xr.receiver_reference_time_report =
true;
// Start with realtime video.
GetVideoEncoderConfig()->content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
// Encoder config for the second part of the test uses screenshare.
encoder_config_with_screenshare = GetVideoEncoderConfig()->Copy();
encoder_config_with_screenshare.content_type =
VideoEncoderConfig::ContentType::kScreen;
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
test.PerformTest();
// Replace old send stream.
SendTask(task_queue(), [this, &encoder_config_with_screenshare]() {
DestroyVideoSendStreams();
CreateVideoSendStream(encoder_config_with_screenshare);
SetVideoDegradation(DegradationPreference::BALANCED);
GetVideoSendStream()->Start();
});
// Continue to run test but now with screenshare.
test.PerformTest();
SendTask(task_queue(), [this]() {
Stop();
DestroyStreams();
DestroyCalls();
});
// Verify that stats have been updated for both screenshare and video.
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayInMs"));
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.Screenshare.InterframeDelayInMs"));
EXPECT_METRIC_EQ(1,
metrics::NumSamples("WebRTC.Video.InterframeDelayMaxInMs"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(
"WebRTC.Video.Screenshare.InterframeDelayMaxInMs"));
}
TEST_F(StatsEndToEndTest, VerifyNackStats) {
static const int kPacketNumberToDrop = 200;
class NackObserver : public test::EndToEndTest {
public:
explicit NackObserver(TaskQueueBase* task_queue)
: EndToEndTest(kLongTimeout), task_queue_(task_queue) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
{
MutexLock lock(&mutex_);
if (++sent_rtp_packets_ == kPacketNumberToDrop) {
RtpPacket header;
EXPECT_TRUE(header.Parse(packet, length));
dropped_rtp_packet_ = header.SequenceNumber();
return DROP_PACKET;
}
}
task_queue_->PostTask(
SafeTask(task_safety_flag_, [this]() { VerifyStats(); }));
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
MutexLock lock(&mutex_);
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
if (!nacks.empty() && absl::c_linear_search(nacks, dropped_rtp_packet_)) {
dropped_rtp_packet_requested_ = true;
}
return SEND_PACKET;
}
void VerifyStats() {
MutexLock lock(&mutex_);
if (!dropped_rtp_packet_requested_)
return;
int send_stream_nack_packets = 0;
int receive_stream_nack_packets = 0;
VideoSendStream::Stats stats = send_stream_->GetStats();
for (const auto& kv : stats.substreams) {
const VideoSendStream::StreamStats& stream_stats = kv.second;
send_stream_nack_packets +=
stream_stats.rtcp_packet_type_counts.nack_packets;
}
for (const auto& receive_stream : receive_streams_) {
VideoReceiveStreamInterface::Stats stats = receive_stream->GetStats();
receive_stream_nack_packets +=
stats.rtcp_packet_type_counts.nack_packets;
}
if (send_stream_nack_packets >= 1 && receive_stream_nack_packets >= 1) {
// NACK packet sent on receive stream and received on sent stream.
if (MinMetricRunTimePassed())
observation_complete_.Set();
}
}
bool MinMetricRunTimePassed() {
int64_t now_ms = Clock::GetRealTimeClock()->TimeInMilliseconds();
if (!start_runtime_ms_)
start_runtime_ms_ = now_ms;
int64_t elapsed_sec = (now_ms - *start_runtime_ms_) / 1000;
return elapsed_sec > metrics::kMinRunTimeInSeconds;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
}
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override {
send_stream_ = send_stream;
receive_streams_ = receive_streams;
}
void OnStreamsStopped() override { task_safety_flag_->SetNotAlive(); }
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
}
Mutex mutex_;
uint64_t sent_rtp_packets_ RTC_GUARDED_BY(&mutex_) = 0;
uint16_t dropped_rtp_packet_ RTC_GUARDED_BY(&mutex_) = 0;
bool dropped_rtp_packet_requested_ RTC_GUARDED_BY(&mutex_) = false;
std::vector<VideoReceiveStreamInterface*> receive_streams_;
VideoSendStream* send_stream_ = nullptr;
absl::optional<int64_t> start_runtime_ms_;
TaskQueueBase* const task_queue_;
rtc::scoped_refptr<PendingTaskSafetyFlag> task_safety_flag_ =
PendingTaskSafetyFlag::CreateDetached();
} test(task_queue());
metrics::Reset();
RunBaseTest(&test);
EXPECT_METRIC_EQ(
1, metrics::NumSamples("WebRTC.Video.UniqueNackRequestsSentInPercent"));
EXPECT_METRIC_EQ(1, metrics::NumSamples(
"WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
EXPECT_METRIC_GT(metrics::MinSample("WebRTC.Video.NackPacketsSentPerMinute"),
0);
}
TEST_F(StatsEndToEndTest, CallReportsRttForSender) {
static const int kSendDelayMs = 30;
static const int kReceiveDelayMs = 70;
SendTask(task_queue(), [this]() {
BuiltInNetworkBehaviorConfig config;
config.queue_delay_ms = kSendDelayMs;
CreateCalls();
CreateSendTransport(config, /*observer*/ nullptr);
config.queue_delay_ms = kReceiveDelayMs;
CreateReceiveTransport(config, /*observer*/ nullptr);
CreateSendConfig(1, 0, 0);
CreateMatchingReceiveConfigs();
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
Start();
});
int64_t start_time_ms = clock_->TimeInMilliseconds();
while (true) {
Call::Stats stats;
SendTask(task_queue(),
[this, &stats]() { stats = sender_call_->GetStats(); });
ASSERT_GE(start_time_ms + kDefaultTimeout.ms(),
clock_->TimeInMilliseconds())
<< "No RTT stats before timeout!";
if (stats.rtt_ms != -1) {
// To avoid failures caused by rounding or minor ntp clock adjustments,
// relax expectation by 1ms.
constexpr int kAllowedErrorMs = 1;
EXPECT_GE(stats.rtt_ms, kSendDelayMs + kReceiveDelayMs - kAllowedErrorMs);
break;
}
SleepMs(10);
}
SendTask(task_queue(), [this]() {
Stop();
DestroyStreams();
DestroyCalls();
});
}
} // namespace webrtc
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