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webrtc / src.git / 0a52f1def68b1f4e90470908cd5dd5105712ff44 / . / video / end_to_end_tests.cc
blob: 6df63eef8b3e4bb9ec5681ea612c82fa781ffae6 [file] [log] [blame]
/*
* Copyright (c) 2013 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 <algorithm>
#include <list>
#include <map>
#include <memory>
#include <sstream>
#include <string>
#include <vector>
#include "api/optional.h"
#include "api/video_codecs/video_encoder.h"
#include "call/call.h"
#include "common_video/include/frame_callback.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "media/base/fakevideorenderer.h"
#include "media/base/mediaconstants.h"
#include "media/engine/internalencoderfactory.h"
#include "media/engine/simulcast_encoder_adapter.h"
#include "media/engine/webrtcvideoencoderfactory.h"
#include "modules/include/module_common_types.h"
#include "modules/rtp_rtcp/include/rtp_rtcp.h"
#include "modules/rtp_rtcp/source/byte_io.h"
#include "modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "modules/rtp_rtcp/source/rtp_format.h"
#include "modules/rtp_rtcp/source/rtp_utility.h"
#include "modules/video_coding/codecs/h264/include/h264.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "modules/video_coding/codecs/vp9/include/vp9.h"
#include "modules/video_coding/include/video_coding_defines.h"
#include "rtc_base/checks.h"
#include "rtc_base/event.h"
#include "rtc_base/file.h"
#include "rtc_base/ptr_util.h"
#include "rtc_base/random.h"
#include "rtc_base/rate_limiter.h"
#include "system_wrappers/include/metrics.h"
#include "system_wrappers/include/metrics_default.h"
#include "system_wrappers/include/sleep.h"
#include "test/call_test.h"
#include "test/direct_transport.h"
#include "test/encoder_settings.h"
#include "test/fake_decoder.h"
#include "test/fake_encoder.h"
#include "test/field_trial.h"
#include "test/frame_generator.h"
#include "test/frame_generator_capturer.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/null_transport.h"
#include "test/rtcp_packet_parser.h"
#include "test/rtp_rtcp_observer.h"
#include "test/testsupport/fileutils.h"
#include "test/testsupport/perf_test.h"
#include "video/transport_adapter.h"
namespace webrtc {
namespace {
constexpr int kSilenceTimeoutMs = 2000;
}
class EndToEndTest : public test::CallTest,
public testing::WithParamInterface<std::string> {
public:
EndToEndTest() : field_trial_(GetParam()) {}
virtual ~EndToEndTest() {
EXPECT_EQ(nullptr, video_send_stream_);
EXPECT_TRUE(video_receive_streams_.empty());
}
protected:
class UnusedTransport : public Transport {
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
ADD_FAILURE() << "Unexpected RTP sent.";
return false;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
ADD_FAILURE() << "Unexpected RTCP sent.";
return false;
}
};
class RequiredTransport : public Transport {
public:
RequiredTransport(bool rtp_required, bool rtcp_required)
: need_rtp_(rtp_required), need_rtcp_(rtcp_required) {}
~RequiredTransport() {
if (need_rtp_) {
ADD_FAILURE() << "Expected RTP packet not sent.";
}
if (need_rtcp_) {
ADD_FAILURE() << "Expected RTCP packet not sent.";
}
}
private:
bool SendRtp(const uint8_t* packet,
size_t length,
const PacketOptions& options) override {
rtc::CritScope lock(&crit_);
need_rtp_ = false;
return true;
}
bool SendRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
need_rtcp_ = false;
return true;
}
bool need_rtp_;
bool need_rtcp_;
rtc::CriticalSection crit_;
};
void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
void ReceivesPliAndRecovers(int rtp_history_ms);
void RespectsRtcpMode(RtcpMode rtcp_mode);
void TestSendsSetSsrcs(size_t num_ssrcs, bool send_single_ssrc_first);
void TestRtpStatePreservation(bool use_rtx, bool provoke_rtcpsr_before_rtp);
void VerifyHistogramStats(bool use_rtx, bool use_fec, bool screenshare);
void VerifyNewVideoSendStreamsRespectNetworkState(
MediaType network_to_bring_up,
VideoEncoder* encoder,
Transport* transport);
void VerifyNewVideoReceiveStreamsRespectNetworkState(
MediaType network_to_bring_up,
Transport* transport);
test::ScopedFieldTrials field_trial_;
};
TEST_P(EndToEndTest, ReceiverCanBeStartedTwice) {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateVideoStreams();
video_receive_streams_[0]->Start();
video_receive_streams_[0]->Start();
DestroyStreams();
}
TEST_P(EndToEndTest, ReceiverCanBeStoppedTwice) {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateVideoStreams();
video_receive_streams_[0]->Stop();
video_receive_streams_[0]->Stop();
DestroyStreams();
}
TEST_P(EndToEndTest, ReceiverCanBeStoppedAndRestarted) {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
test::NullTransport transport;
CreateSendConfig(1, 0, 0, &transport);
CreateMatchingReceiveConfigs(&transport);
CreateVideoStreams();
video_receive_streams_[0]->Stop();
video_receive_streams_[0]->Start();
video_receive_streams_[0]->Stop();
DestroyStreams();
}
TEST_P(EndToEndTest, RendersSingleDelayedFrame) {
static const int kWidth = 320;
static const int kHeight = 240;
// This constant is chosen to be higher than the timeout in the video_render
// module. This makes sure that frames aren't dropped if there are no other
// frames in the queue.
static const int kRenderDelayMs = 1000;
class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
public:
Renderer() : event_(false, false) {}
void OnFrame(const VideoFrame& video_frame) override {
SleepMs(kRenderDelayMs);
event_.Set();
}
bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
rtc::Event event_;
} renderer;
test::FrameForwarder frame_forwarder;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &renderer, &frame_forwarder, &sender_transport,
&receiver_transport]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_receive_configs_[0].renderer = &renderer;
CreateVideoStreams();
Start();
// Create frames that are smaller than the send width/height, this is done
// to check that the callbacks are done after processing video.
std::unique_ptr<test::FrameGenerator> frame_generator(
test::FrameGenerator::CreateSquareGenerator(kWidth, kHeight));
video_send_stream_->SetSource(
&frame_forwarder,
VideoSendStream::DegradationPreference::kMaintainFramerate);
frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(renderer.Wait())
<< "Timed out while waiting for the frame to render.";
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
TEST_P(EndToEndTest, TransmitsFirstFrame) {
class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
public:
Renderer() : event_(false, false) {}
void OnFrame(const VideoFrame& video_frame) override { event_.Set(); }
bool Wait() { return event_.Wait(kDefaultTimeoutMs); }
rtc::Event event_;
} renderer;
std::unique_ptr<test::FrameGenerator> frame_generator;
test::FrameForwarder frame_forwarder;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &renderer, &frame_generator, &frame_forwarder,
&sender_transport, &receiver_transport]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_receive_configs_[0].renderer = &renderer;
CreateVideoStreams();
Start();
frame_generator = test::FrameGenerator::CreateSquareGenerator(
kDefaultWidth, kDefaultHeight);
video_send_stream_->SetSource(
&frame_forwarder,
VideoSendStream::DegradationPreference::kMaintainFramerate);
frame_forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(renderer.Wait())
<< "Timed out while waiting for the frame to render.";
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
class CodecObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
CodecObserver(int no_frames_to_wait_for,
VideoRotation rotation_to_test,
const std::string& payload_name,
std::unique_ptr<webrtc::VideoEncoder> encoder,
std::unique_ptr<webrtc::VideoDecoder> decoder)
: EndToEndTest(4 * webrtc::EndToEndTest::kDefaultTimeoutMs),
// TODO(hta): This timeout (120 seconds) is excessive.
// https://bugs.webrtc.org/6830
no_frames_to_wait_for_(no_frames_to_wait_for),
expected_rotation_(rotation_to_test),
payload_name_(payload_name),
encoder_(std::move(encoder)),
decoder_(std::move(decoder)),
frame_counter_(0) {}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for enough frames to be decoded.";
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = encoder_.get();
send_config->encoder_settings.payload_name = payload_name_;
send_config->encoder_settings.payload_type =
test::CallTest::kVideoSendPayloadType;
(*receive_configs)[0].renderer = this;
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->encoder_settings.payload_type;
(*receive_configs)[0].decoders[0].payload_name =
send_config->encoder_settings.payload_name;
(*receive_configs)[0].decoders[0].decoder = decoder_.get();
}
void OnFrame(const VideoFrame& video_frame) override {
EXPECT_EQ(expected_rotation_, video_frame.rotation());
if (++frame_counter_ == no_frames_to_wait_for_)
observation_complete_.Set();
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->SetFakeRotation(expected_rotation_);
}
private:
int no_frames_to_wait_for_;
VideoRotation expected_rotation_;
std::string payload_name_;
std::unique_ptr<webrtc::VideoEncoder> encoder_;
std::unique_ptr<webrtc::VideoDecoder> decoder_;
int frame_counter_;
};
TEST_P(EndToEndTest, SendsAndReceivesVP8) {
CodecObserver test(5, kVideoRotation_0, "VP8", VP8Encoder::Create(),
VP8Decoder::Create());
RunBaseTest(&test);
}
TEST_P(EndToEndTest, SendsAndReceivesVP8Rotation90) {
CodecObserver test(5, kVideoRotation_90, "VP8", VP8Encoder::Create(),
VP8Decoder::Create());
RunBaseTest(&test);
}
#if !defined(RTC_DISABLE_VP9)
TEST_P(EndToEndTest, SendsAndReceivesVP9) {
CodecObserver test(500, kVideoRotation_0, "VP9", VP9Encoder::Create(),
VP9Decoder::Create());
RunBaseTest(&test);
}
TEST_P(EndToEndTest, SendsAndReceivesVP9VideoRotation90) {
CodecObserver test(5, kVideoRotation_90, "VP9", VP9Encoder::Create(),
VP9Decoder::Create());
RunBaseTest(&test);
}
#endif // !defined(RTC_DISABLE_VP9)
#if defined(WEBRTC_USE_H264)
class EndToEndTestH264 : public EndToEndTest {};
const auto h264_field_trial_combinations = ::testing::Values(
"WebRTC-SpsPpsIdrIsH264Keyframe/Disabled/WebRTC-RoundRobinPacing/Disabled/",
"WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/WebRTC-RoundRobinPacing/Disabled/",
"WebRTC-SpsPpsIdrIsH264Keyframe/Disabled/WebRTC-RoundRobinPacing/Enabled/",
"WebRTC-SpsPpsIdrIsH264Keyframe/Enabled/WebRTC-RoundRobinPacing/Enabled/");
INSTANTIATE_TEST_CASE_P(SpsPpsIdrIsKeyframe,
EndToEndTestH264,
h264_field_trial_combinations);
TEST_P(EndToEndTestH264, SendsAndReceivesH264) {
CodecObserver test(500, kVideoRotation_0, "H264",
H264Encoder::Create(cricket::VideoCodec("H264")),
H264Decoder::Create());
RunBaseTest(&test);
}
TEST_P(EndToEndTestH264, SendsAndReceivesH264VideoRotation90) {
CodecObserver test(5, kVideoRotation_90, "H264",
H264Encoder::Create(cricket::VideoCodec("H264")),
H264Decoder::Create());
RunBaseTest(&test);
}
TEST_P(EndToEndTestH264, SendsAndReceivesH264PacketizationMode0) {
cricket::VideoCodec codec = cricket::VideoCodec("H264");
codec.SetParam(cricket::kH264FmtpPacketizationMode, "0");
CodecObserver test(500, kVideoRotation_0, "H264", H264Encoder::Create(codec),
H264Decoder::Create());
RunBaseTest(&test);
}
TEST_P(EndToEndTestH264, SendsAndReceivesH264PacketizationMode1) {
cricket::VideoCodec codec = cricket::VideoCodec("H264");
codec.SetParam(cricket::kH264FmtpPacketizationMode, "1");
CodecObserver test(500, kVideoRotation_0, "H264", H264Encoder::Create(codec),
H264Decoder::Create());
RunBaseTest(&test);
}
#endif // defined(WEBRTC_USE_H264)
TEST_P(EndToEndTest, ReceiverUsesLocalSsrc) {
class SyncRtcpObserver : public test::EndToEndTest {
public:
SyncRtcpObserver() : EndToEndTest(kDefaultTimeoutMs) {}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
EXPECT_EQ(kReceiverLocalVideoSsrc, parser.sender_ssrc());
observation_complete_.Set();
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for a receiver RTCP packet to be sent.";
}
} test;
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ReceivesAndRetransmitsNack) {
static const int kNumberOfNacksToObserve = 2;
static const int kLossBurstSize = 2;
static const int kPacketsBetweenLossBursts = 9;
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(kLongTimeoutMs),
sent_rtp_packets_(0),
packets_left_to_drop_(0),
nacks_left_(kNumberOfNacksToObserve) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
// Never drop retransmitted packets.
if (dropped_packets_.find(header.sequenceNumber) !=
dropped_packets_.end()) {
retransmitted_packets_.insert(header.sequenceNumber);
return SEND_PACKET;
}
if (nacks_left_ <= 0 &&
retransmitted_packets_.size() == dropped_packets_.size()) {
observation_complete_.Set();
}
++sent_rtp_packets_;
// Enough NACKs received, stop dropping packets.
if (nacks_left_ <= 0)
return SEND_PACKET;
// Check if it's time for a new loss burst.
if (sent_rtp_packets_ % kPacketsBetweenLossBursts == 0)
packets_left_to_drop_ = kLossBurstSize;
// Never drop padding packets as those won't be retransmitted.
if (packets_left_to_drop_ > 0 && header.paddingLength == 0) {
--packets_left_to_drop_;
dropped_packets_.insert(header.sequenceNumber);
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
nacks_left_ -= parser.nack()->num_packets();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::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 PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out waiting for packets to be NACKed, retransmitted and "
"rendered.";
}
rtc::CriticalSection crit_;
std::set<uint16_t> dropped_packets_;
std::set<uint16_t> retransmitted_packets_;
uint64_t sent_rtp_packets_;
int packets_left_to_drop_;
int nacks_left_ RTC_GUARDED_BY(&crit_);
} test;
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ReceivesNackAndRetransmitsAudio) {
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(kLongTimeoutMs),
local_ssrc_(0),
remote_ssrc_(0),
receive_transport_(nullptr) {}
private:
size_t GetNumVideoStreams() const override { return 0; }
size_t GetNumAudioStreams() const override { return 1; }
test::PacketTransport* CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue) override {
test::PacketTransport* receive_transport = new test::PacketTransport(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
payload_type_map_, FakeNetworkPipe::Config());
receive_transport_ = receive_transport;
return receive_transport;
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
if (!sequence_number_to_retransmit_) {
sequence_number_to_retransmit_ = header.sequenceNumber;
// Don't ask for retransmission straight away, may be deduped in pacer.
} else if (header.sequenceNumber == *sequence_number_to_retransmit_) {
observation_complete_.Set();
} else {
// Send a NACK as often as necessary until retransmission is received.
rtcp::Nack nack;
nack.SetSenderSsrc(local_ssrc_);
nack.SetMediaSsrc(remote_ssrc_);
uint16_t nack_list[] = {*sequence_number_to_retransmit_};
nack.SetPacketIds(nack_list, 1);
rtc::Buffer buffer = nack.Build();
EXPECT_TRUE(receive_transport_->SendRtcp(buffer.data(), buffer.size()));
}
return SEND_PACKET;
}
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out waiting for packets to be NACKed, retransmitted and "
"rendered.";
}
uint32_t local_ssrc_;
uint32_t remote_ssrc_;
Transport* receive_transport_;
rtc::Optional<uint16_t> sequence_number_to_retransmit_;
} test;
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ReceivesUlpfec) {
class UlpfecRenderObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
UlpfecRenderObserver()
: EndToEndTest(kDefaultTimeoutMs),
encoder_(VP8Encoder::Create()),
random_(0xcafef00d1),
num_packets_sent_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
EXPECT_TRUE(header.payloadType == kVideoSendPayloadType ||
header.payloadType == kRedPayloadType)
<< "Unknown payload type received.";
EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc) << "Unknown SSRC received.";
// Parse RED header.
int encapsulated_payload_type = -1;
if (header.payloadType == kRedPayloadType) {
encapsulated_payload_type =
static_cast<int>(packet[header.headerLength]);
EXPECT_TRUE(encapsulated_payload_type == kVideoSendPayloadType ||
encapsulated_payload_type == kUlpfecPayloadType)
<< "Unknown encapsulated payload type received.";
}
// To minimize test flakiness, always let ULPFEC packets through.
if (encapsulated_payload_type == kUlpfecPayloadType) {
return SEND_PACKET;
}
// Simulate 5% video packet loss after rampup period. Record the
// corresponding timestamps that were dropped.
if (num_packets_sent_++ > 100 && random_.Rand(1, 100) <= 5) {
if (encapsulated_payload_type == kVideoSendPayloadType) {
dropped_sequence_numbers_.insert(header.sequenceNumber);
dropped_timestamps_.insert(header.timestamp);
}
return DROP_PACKET;
}
return SEND_PACKET;
}
void OnFrame(const VideoFrame& video_frame) override {
rtc::CritScope lock(&crit_);
// Rendering frame with timestamp of packet that was dropped -> FEC
// protection worked.
auto it = dropped_timestamps_.find(video_frame.timestamp());
if (it != dropped_timestamps_.end()) {
observation_complete_.Set();
}
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Use VP8 instead of FAKE, since the latter does not have PictureID
// in the packetization headers.
send_config->encoder_settings.encoder = encoder_.get();
send_config->encoder_settings.payload_name = "VP8";
send_config->encoder_settings.payload_type = kVideoSendPayloadType;
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config->encoder_settings);
decoder_.reset(decoder.decoder);
(*receive_configs)[0].decoders.clear();
(*receive_configs)[0].decoders.push_back(decoder);
// Enable ULPFEC over RED.
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
(*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
(*receive_configs)[0].renderer = this;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out waiting for dropped frames to be rendered.";
}
rtc::CriticalSection crit_;
std::unique_ptr<VideoEncoder> encoder_;
std::unique_ptr<VideoDecoder> decoder_;
std::set<uint32_t> dropped_sequence_numbers_ RTC_GUARDED_BY(crit_);
// Several packets can have the same timestamp.
std::multiset<uint32_t> dropped_timestamps_ RTC_GUARDED_BY(crit_);
Random random_;
int num_packets_sent_ RTC_GUARDED_BY(crit_);
} test;
RunBaseTest(&test);
}
class FlexfecRenderObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
static constexpr uint32_t kVideoLocalSsrc = 123;
static constexpr uint32_t kFlexfecLocalSsrc = 456;
explicit FlexfecRenderObserver(bool enable_nack, bool expect_flexfec_rtcp)
: test::EndToEndTest(test::CallTest::kDefaultTimeoutMs),
enable_nack_(enable_nack),
expect_flexfec_rtcp_(expect_flexfec_rtcp),
received_flexfec_rtcp_(false),
random_(0xcafef00d1),
num_packets_sent_(0) {}
size_t GetNumFlexfecStreams() const override { return 1; }
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
EXPECT_TRUE(header.payloadType ==
test::CallTest::kFakeVideoSendPayloadType ||
header.payloadType == test::CallTest::kFlexfecPayloadType ||
(enable_nack_ &&
header.payloadType == test::CallTest::kSendRtxPayloadType))
<< "Unknown payload type received.";
EXPECT_TRUE(
header.ssrc == test::CallTest::kVideoSendSsrcs[0] ||
header.ssrc == test::CallTest::kFlexfecSendSsrc ||
(enable_nack_ && header.ssrc == test::CallTest::kSendRtxSsrcs[0]))
<< "Unknown SSRC received.";
// To reduce test flakiness, always let FlexFEC packets through.
if (header.payloadType == test::CallTest::kFlexfecPayloadType) {
EXPECT_EQ(test::CallTest::kFlexfecSendSsrc, header.ssrc);
return SEND_PACKET;
}
// To reduce test flakiness, always let RTX packets through.
if (header.payloadType == test::CallTest::kSendRtxPayloadType) {
EXPECT_EQ(test::CallTest::kSendRtxSsrcs[0], header.ssrc);
// Parse RTX header.
uint16_t original_sequence_number =
ByteReader<uint16_t>::ReadBigEndian(&packet[header.headerLength]);
// From the perspective of FEC, a retransmitted packet is no longer
// dropped, so remove it from list of dropped packets.
auto seq_num_it =
dropped_sequence_numbers_.find(original_sequence_number);
if (seq_num_it != dropped_sequence_numbers_.end()) {
dropped_sequence_numbers_.erase(seq_num_it);
auto ts_it = dropped_timestamps_.find(header.timestamp);
EXPECT_NE(ts_it, dropped_timestamps_.end());
dropped_timestamps_.erase(ts_it);
}
return SEND_PACKET;
}
// Simulate 5% video packet loss after rampup period. Record the
// corresponding timestamps that were dropped.
if (num_packets_sent_++ > 100 && random_.Rand(1, 100) <= 5) {
EXPECT_EQ(test::CallTest::kFakeVideoSendPayloadType, header.payloadType);
EXPECT_EQ(test::CallTest::kVideoSendSsrcs[0], header.ssrc);
dropped_sequence_numbers_.insert(header.sequenceNumber);
dropped_timestamps_.insert(header.timestamp);
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
test::RtcpPacketParser parser;
parser.Parse(data, length);
if (parser.sender_ssrc() == kFlexfecLocalSsrc) {
EXPECT_EQ(1, parser.receiver_report()->num_packets());
const std::vector<rtcp::ReportBlock>& report_blocks =
parser.receiver_report()->report_blocks();
if (!report_blocks.empty()) {
EXPECT_EQ(1U, report_blocks.size());
EXPECT_EQ(test::CallTest::kFlexfecSendSsrc,
report_blocks[0].source_ssrc());
rtc::CritScope lock(&crit_);
received_flexfec_rtcp_ = true;
}
}
return SEND_PACKET;
}
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
// At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
const int kNetworkDelayMs = 100;
FakeNetworkPipe::Config config;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(task_queue, sender_call, this,
test::PacketTransport::kSender,
test::CallTest::payload_type_map_, config);
}
void OnFrame(const VideoFrame& video_frame) override {
EXPECT_EQ(kVideoRotation_90, video_frame.rotation());
rtc::CritScope lock(&crit_);
// Rendering frame with timestamp of packet that was dropped -> FEC
// protection worked.
auto it = dropped_timestamps_.find(video_frame.timestamp());
if (it != dropped_timestamps_.end()) {
if (!expect_flexfec_rtcp_ || received_flexfec_rtcp_) {
observation_complete_.Set();
}
}
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].rtp.local_ssrc = kVideoLocalSsrc;
(*receive_configs)[0].renderer = this;
if (enable_nack_) {
send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
send_config->rtp.rtx.ssrcs.push_back(test::CallTest::kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
(*receive_configs)[0].rtp.nack.rtp_history_ms =
test::CallTest::kNackRtpHistoryMs;
(*receive_configs)[0].rtp.rtx_ssrc = test::CallTest::kSendRtxSsrcs[0];
(*receive_configs)[0]
.rtp
.rtx_associated_payload_types[test::CallTest::kSendRtxPayloadType] =
test::CallTest::kVideoSendPayloadType;
}
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
}
void ModifyFlexfecConfigs(
std::vector<FlexfecReceiveStream::Config>* receive_configs) override {
(*receive_configs)[0].local_ssrc = kFlexfecLocalSsrc;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out waiting for dropped frames to be rendered.";
}
rtc::CriticalSection crit_;
std::set<uint32_t> dropped_sequence_numbers_ RTC_GUARDED_BY(crit_);
// Several packets can have the same timestamp.
std::multiset<uint32_t> dropped_timestamps_ RTC_GUARDED_BY(crit_);
const bool enable_nack_;
const bool expect_flexfec_rtcp_;
bool received_flexfec_rtcp_ RTC_GUARDED_BY(crit_);
Random random_;
int num_packets_sent_;
};
TEST_P(EndToEndTest, RecoversWithFlexfec) {
FlexfecRenderObserver test(false, false);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, RecoversWithFlexfecAndNack) {
FlexfecRenderObserver test(true, false);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, RecoversWithFlexfecAndSendsCorrespondingRtcp) {
FlexfecRenderObserver test(false, true);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ReceivedUlpfecPacketsNotNacked) {
class UlpfecNackObserver : public test::EndToEndTest {
public:
UlpfecNackObserver()
: EndToEndTest(kDefaultTimeoutMs),
state_(kFirstPacket),
ulpfec_sequence_number_(0),
has_last_sequence_number_(false),
last_sequence_number_(0),
encoder_(VP8Encoder::Create()),
decoder_(VP8Decoder::Create()) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock_(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
int encapsulated_payload_type = -1;
if (header.payloadType == kRedPayloadType) {
encapsulated_payload_type =
static_cast<int>(packet[header.headerLength]);
if (encapsulated_payload_type != kFakeVideoSendPayloadType)
EXPECT_EQ(kUlpfecPayloadType, encapsulated_payload_type);
} else {
EXPECT_EQ(kFakeVideoSendPayloadType, header.payloadType);
}
if (has_last_sequence_number_ &&
!IsNewerSequenceNumber(header.sequenceNumber,
last_sequence_number_)) {
// Drop retransmitted packets.
return DROP_PACKET;
}
last_sequence_number_ = header.sequenceNumber;
has_last_sequence_number_ = true;
bool ulpfec_packet = encapsulated_payload_type == kUlpfecPayloadType;
switch (state_) {
case kFirstPacket:
state_ = kDropEveryOtherPacketUntilUlpfec;
break;
case kDropEveryOtherPacketUntilUlpfec:
if (ulpfec_packet) {
state_ = kDropAllMediaPacketsUntilUlpfec;
} else if (header.sequenceNumber % 2 == 0) {
return DROP_PACKET;
}
break;
case kDropAllMediaPacketsUntilUlpfec:
if (!ulpfec_packet)
return DROP_PACKET;
ulpfec_sequence_number_ = header.sequenceNumber;
state_ = kDropOneMediaPacket;
break;
case kDropOneMediaPacket:
if (ulpfec_packet)
return DROP_PACKET;
state_ = kPassOneMediaPacket;
return DROP_PACKET;
break;
case kPassOneMediaPacket:
if (ulpfec_packet)
return DROP_PACKET;
// Pass one media packet after dropped packet after last FEC,
// otherwise receiver might never see a seq_no after
// |ulpfec_sequence_number_|
state_ = kVerifyUlpfecPacketNotInNackList;
break;
case kVerifyUlpfecPacketNotInNackList:
// Continue to drop packets. Make sure no frame can be decoded.
if (ulpfec_packet || header.sequenceNumber % 2 == 0)
return DROP_PACKET;
break;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock_(&crit_);
if (state_ == kVerifyUlpfecPacketNotInNackList) {
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
EXPECT_TRUE(std::find(nacks.begin(), nacks.end(),
ulpfec_sequence_number_) == nacks.end())
<< "Got nack for ULPFEC packet";
if (!nacks.empty() &&
IsNewerSequenceNumber(nacks.back(), ulpfec_sequence_number_)) {
observation_complete_.Set();
}
}
return SEND_PACKET;
}
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
// At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
// Configure some network delay.
const int kNetworkDelayMs = 50;
FakeNetworkPipe::Config config;
config.queue_delay_ms = kNetworkDelayMs;
return new test::PacketTransport(task_queue, sender_call, this,
test::PacketTransport::kSender,
payload_type_map_, config);
}
// TODO(holmer): Investigate why we don't send FEC packets when the bitrate
// is 10 kbps.
Call::Config GetSenderCallConfig() override {
Call::Config config(event_log_.get());
const int kMinBitrateBps = 30000;
config.bitrate_config.min_bitrate_bps = kMinBitrateBps;
return config;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Configure hybrid NACK/FEC.
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
// Set codec to VP8, otherwise NACK/FEC hybrid will be disabled.
send_config->encoder_settings.encoder = encoder_.get();
send_config->encoder_settings.payload_name = "VP8";
send_config->encoder_settings.payload_type = kFakeVideoSendPayloadType;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.ulpfec_payload_type = kUlpfecPayloadType;
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->encoder_settings.payload_type;
(*receive_configs)[0].decoders[0].payload_name =
send_config->encoder_settings.payload_name;
(*receive_configs)[0].decoders[0].decoder = decoder_.get();
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for FEC packets to be received.";
}
enum {
kFirstPacket,
kDropEveryOtherPacketUntilUlpfec,
kDropAllMediaPacketsUntilUlpfec,
kDropOneMediaPacket,
kPassOneMediaPacket,
kVerifyUlpfecPacketNotInNackList,
} state_;
rtc::CriticalSection crit_;
uint16_t ulpfec_sequence_number_ RTC_GUARDED_BY(&crit_);
bool has_last_sequence_number_;
uint16_t last_sequence_number_;
std::unique_ptr<webrtc::VideoEncoder> encoder_;
std::unique_ptr<webrtc::VideoDecoder> decoder_;
} test;
RunBaseTest(&test);
}
// This test drops second RTP packet with a marker bit set, makes sure it's
// retransmitted and renders. Retransmission SSRCs are also checked.
void EndToEndTest::DecodesRetransmittedFrame(bool enable_rtx, bool enable_red) {
static const int kDroppedFrameNumber = 10;
class RetransmissionObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
RetransmissionObserver(bool enable_rtx, bool enable_red)
: EndToEndTest(kDefaultTimeoutMs),
payload_type_(GetPayloadType(false, enable_red)),
retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0]
: kVideoSendSsrcs[0]),
retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
encoder_(VP8Encoder::Create()),
marker_bits_observed_(0),
retransmitted_timestamp_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
// Ignore padding-only packets over RTX.
if (header.payloadType != payload_type_) {
EXPECT_EQ(retransmission_ssrc_, header.ssrc);
if (length == header.headerLength + header.paddingLength)
return SEND_PACKET;
}
if (header.timestamp == retransmitted_timestamp_) {
EXPECT_EQ(retransmission_ssrc_, header.ssrc);
EXPECT_EQ(retransmission_payload_type_, header.payloadType);
return SEND_PACKET;
}
// Found the final packet of the frame to inflict loss to, drop this and
// expect a retransmission.
if (header.payloadType == payload_type_ && header.markerBit &&
++marker_bits_observed_ == kDroppedFrameNumber) {
// This should be the only dropped packet.
EXPECT_EQ(0u, retransmitted_timestamp_);
retransmitted_timestamp_ = header.timestamp;
if (std::find(rendered_timestamps_.begin(), rendered_timestamps_.end(),
retransmitted_timestamp_) != rendered_timestamps_.end()) {
// Frame was rendered before last packet was scheduled for sending.
// This is extremly rare but possible scenario because prober able to
// resend packet before it was send.
// TODO(danilchap): Remove this corner case when prober would not be
// able to sneak in between packet saved to history for resending and
// pacer notified about existance of that packet for sending.
// See https://bugs.chromium.org/p/webrtc/issues/detail?id=5540 for
// details.
observation_complete_.Set();
}
return DROP_PACKET;
}
return SEND_PACKET;
}
void OnFrame(const VideoFrame& frame) override {
EXPECT_EQ(kVideoRotation_90, frame.rotation());
{
rtc::CritScope lock(&crit_);
if (frame.timestamp() == retransmitted_timestamp_)
observation_complete_.Set();
rendered_timestamps_.push_back(frame.timestamp());
}
orig_renderer_->OnFrame(frame);
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
// Insert ourselves into the rendering pipeline.
RTC_DCHECK(!orig_renderer_);
orig_renderer_ = (*receive_configs)[0].renderer;
RTC_DCHECK(orig_renderer_);
(*receive_configs)[0].disable_prerenderer_smoothing = true;
(*receive_configs)[0].renderer = this;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
if (payload_type_ == kRedPayloadType) {
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
if (retransmission_ssrc_ == kSendRtxSsrcs[0])
send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
(*receive_configs)[0].rtp.ulpfec_payload_type =
send_config->rtp.ulpfec.ulpfec_payload_type;
(*receive_configs)[0].rtp.red_payload_type =
send_config->rtp.ulpfec.red_payload_type;
}
if (retransmission_ssrc_ == kSendRtxSsrcs[0]) {
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
(*receive_configs)[0]
.rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType)
? kRtxRedPayloadType
: kSendRtxPayloadType] =
payload_type_;
}
// Configure encoding and decoding with VP8, since generic packetization
// doesn't support FEC with NACK.
RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size());
send_config->encoder_settings.encoder = encoder_.get();
send_config->encoder_settings.payload_name = "VP8";
(*receive_configs)[0].decoders[0].payload_name = "VP8";
}
void OnFrameGeneratorCapturerCreated(
test::FrameGeneratorCapturer* frame_generator_capturer) override {
frame_generator_capturer->SetFakeRotation(kVideoRotation_90);
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for retransmission to render.";
}
int GetPayloadType(bool use_rtx, bool use_fec) {
if (use_fec) {
if (use_rtx)
return kRtxRedPayloadType;
return kRedPayloadType;
}
if (use_rtx)
return kSendRtxPayloadType;
return kFakeVideoSendPayloadType;
}
rtc::CriticalSection crit_;
rtc::VideoSinkInterface<VideoFrame>* orig_renderer_ = nullptr;
const int payload_type_;
const uint32_t retransmission_ssrc_;
const int retransmission_payload_type_;
std::unique_ptr<VideoEncoder> encoder_;
const std::string payload_name_;
int marker_bits_observed_;
uint32_t retransmitted_timestamp_ RTC_GUARDED_BY(&crit_);
std::vector<uint32_t> rendered_timestamps_ RTC_GUARDED_BY(&crit_);
} test(enable_rtx, enable_red);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, DecodesRetransmittedFrame) {
DecodesRetransmittedFrame(false, false);
}
TEST_P(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
DecodesRetransmittedFrame(true, false);
}
TEST_P(EndToEndTest, DecodesRetransmittedFrameByRed) {
DecodesRetransmittedFrame(false, true);
}
TEST_P(EndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
DecodesRetransmittedFrame(true, true);
}
void EndToEndTest::ReceivesPliAndRecovers(int rtp_history_ms) {
static const int kPacketsToDrop = 1;
class PliObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
explicit PliObserver(int rtp_history_ms)
: EndToEndTest(kLongTimeoutMs),
rtp_history_ms_(rtp_history_ms),
nack_enabled_(rtp_history_ms > 0),
highest_dropped_timestamp_(0),
frames_to_drop_(0),
received_pli_(false) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
// Drop all retransmitted packets to force a PLI.
if (header.timestamp <= highest_dropped_timestamp_)
return DROP_PACKET;
if (frames_to_drop_ > 0) {
highest_dropped_timestamp_ = header.timestamp;
--frames_to_drop_;
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
if (!nack_enabled_)
EXPECT_EQ(0, parser.nack()->num_packets());
if (parser.pli()->num_packets() > 0)
received_pli_ = true;
return SEND_PACKET;
}
void OnFrame(const VideoFrame& video_frame) override {
rtc::CritScope lock(&crit_);
if (received_pli_ &&
video_frame.timestamp() > highest_dropped_timestamp_) {
observation_complete_.Set();
}
if (!received_pli_)
frames_to_drop_ = kPacketsToDrop;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = rtp_history_ms_;
(*receive_configs)[0].rtp.nack.rtp_history_ms = rtp_history_ms_;
(*receive_configs)[0].renderer = this;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for PLI to be "
"received and a frame to be "
"rendered afterwards.";
}
rtc::CriticalSection crit_;
int rtp_history_ms_;
bool nack_enabled_;
uint32_t highest_dropped_timestamp_ RTC_GUARDED_BY(&crit_);
int frames_to_drop_ RTC_GUARDED_BY(&crit_);
bool received_pli_ RTC_GUARDED_BY(&crit_);
} test(rtp_history_ms);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ReceivesPliAndRecoversWithNack) {
ReceivesPliAndRecovers(1000);
}
TEST_P(EndToEndTest, ReceivesPliAndRecoversWithoutNack) {
ReceivesPliAndRecovers(0);
}
TEST_P(EndToEndTest, UnknownRtpPacketGivesUnknownSsrcReturnCode) {
class PacketInputObserver : public PacketReceiver {
public:
explicit PacketInputObserver(PacketReceiver* receiver)
: receiver_(receiver), delivered_packet_(false, false) {}
bool Wait() { return delivered_packet_.Wait(kDefaultTimeoutMs); }
private:
DeliveryStatus DeliverPacket(MediaType media_type,
rtc::CopyOnWriteBuffer packet,
const PacketTime& packet_time) override {
if (RtpHeaderParser::IsRtcp(packet.cdata(), packet.size())) {
return receiver_->DeliverPacket(media_type, std::move(packet),
packet_time);
} else {
DeliveryStatus delivery_status = receiver_->DeliverPacket(
media_type, std::move(packet), packet_time);
EXPECT_EQ(DELIVERY_UNKNOWN_SSRC, delivery_status);
delivered_packet_.Set();
return delivery_status;
}
}
PacketReceiver* receiver_;
rtc::Event delivered_packet_;
};
std::unique_ptr<test::DirectTransport> send_transport;
std::unique_ptr<test::DirectTransport> receive_transport;
std::unique_ptr<PacketInputObserver> input_observer;
task_queue_.SendTask([this, &send_transport, &receive_transport,
&input_observer]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
send_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receive_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
input_observer =
rtc::MakeUnique<PacketInputObserver>(receiver_call_->Receiver());
send_transport->SetReceiver(input_observer.get());
receive_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, send_transport.get());
CreateMatchingReceiveConfigs(receive_transport.get());
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
receiver_call_->DestroyVideoReceiveStream(video_receive_streams_[0]);
video_receive_streams_.clear();
});
// Wait() waits for a received packet.
EXPECT_TRUE(input_observer->Wait());
task_queue_.SendTask([this, &send_transport, &receive_transport]() {
Stop();
DestroyStreams();
send_transport.reset();
receive_transport.reset();
DestroyCalls();
});
}
void EndToEndTest::RespectsRtcpMode(RtcpMode rtcp_mode) {
static const int kNumCompoundRtcpPacketsToObserve = 10;
class RtcpModeObserver : public test::EndToEndTest {
public:
explicit RtcpModeObserver(RtcpMode rtcp_mode)
: EndToEndTest(kDefaultTimeoutMs),
rtcp_mode_(rtcp_mode),
sent_rtp_(0),
sent_rtcp_(0) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
if (++sent_rtp_ % 3 == 0)
return DROP_PACKET;
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
++sent_rtcp_;
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
EXPECT_EQ(0, parser.sender_report()->num_packets());
switch (rtcp_mode_) {
case RtcpMode::kCompound:
// TODO(holmer): We shouldn't send transport feedback alone if
// compound RTCP is negotiated.
if (parser.receiver_report()->num_packets() == 0 &&
parser.transport_feedback()->num_packets() == 0) {
ADD_FAILURE() << "Received RTCP packet without receiver report for "
"RtcpMode::kCompound.";
observation_complete_.Set();
}
if (sent_rtcp_ >= kNumCompoundRtcpPacketsToObserve)
observation_complete_.Set();
break;
case RtcpMode::kReducedSize:
if (parser.receiver_report()->num_packets() == 0)
observation_complete_.Set();
break;
case RtcpMode::kOff:
RTC_NOTREACHED();
break;
}
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.rtcp_mode = rtcp_mode_;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< (rtcp_mode_ == RtcpMode::kCompound
? "Timed out before observing enough compound packets."
: "Timed out before receiving a non-compound RTCP packet.");
}
RtcpMode rtcp_mode_;
rtc::CriticalSection crit_;
// Must be protected since RTCP can be sent by both the process thread
// and the pacer thread.
int sent_rtp_ RTC_GUARDED_BY(&crit_);
int sent_rtcp_ RTC_GUARDED_BY(&crit_);
} test(rtcp_mode);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, UsesRtcpCompoundMode) {
RespectsRtcpMode(RtcpMode::kCompound);
}
TEST_P(EndToEndTest, UsesRtcpReducedSizeMode) {
RespectsRtcpMode(RtcpMode::kReducedSize);
}
// Test sets up a Call multiple senders with different resolutions and SSRCs.
// Another is set up to receive all three of these with different renderers.
class MultiStreamTest {
public:
static constexpr size_t kNumStreams = 3;
const uint8_t kVideoPayloadType = 124;
const std::map<uint8_t, MediaType> payload_type_map_ = {
{kVideoPayloadType, MediaType::VIDEO}};
struct CodecSettings {
uint32_t ssrc;
int width;
int height;
} codec_settings[kNumStreams];
explicit MultiStreamTest(test::SingleThreadedTaskQueueForTesting* task_queue)
: task_queue_(task_queue) {
// TODO(sprang): Cleanup when msvc supports explicit initializers for array.
codec_settings[0] = {1, 640, 480};
codec_settings[1] = {2, 320, 240};
codec_settings[2] = {3, 240, 160};
}
virtual ~MultiStreamTest() {}
void RunTest() {
webrtc::RtcEventLogNullImpl event_log;
Call::Config config(&event_log);
std::unique_ptr<Call> sender_call;
std::unique_ptr<Call> receiver_call;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
VideoSendStream* send_streams[kNumStreams];
VideoReceiveStream* receive_streams[kNumStreams];
test::FrameGeneratorCapturer* frame_generators[kNumStreams];
std::vector<std::unique_ptr<VideoDecoder>> allocated_decoders;
std::unique_ptr<VideoEncoder> encoders[kNumStreams];
task_queue_->SendTask([&]() {
sender_call = rtc::WrapUnique(Call::Create(config));
receiver_call = rtc::WrapUnique(Call::Create(config));
sender_transport =
rtc::WrapUnique(CreateSendTransport(task_queue_, sender_call.get()));
receiver_transport = rtc::WrapUnique(
CreateReceiveTransport(task_queue_, receiver_call.get()));
sender_transport->SetReceiver(receiver_call->Receiver());
receiver_transport->SetReceiver(sender_call->Receiver());
for (size_t i = 0; i < kNumStreams; ++i)
encoders[i] = VP8Encoder::Create();
for (size_t i = 0; i < kNumStreams; ++i) {
uint32_t ssrc = codec_settings[i].ssrc;
int width = codec_settings[i].width;
int height = codec_settings[i].height;
VideoSendStream::Config send_config(sender_transport.get());
send_config.rtp.ssrcs.push_back(ssrc);
send_config.encoder_settings.encoder = encoders[i].get();
send_config.encoder_settings.payload_name = "VP8";
send_config.encoder_settings.payload_type = kVideoPayloadType;
VideoEncoderConfig encoder_config;
test::FillEncoderConfiguration(1, &encoder_config);
encoder_config.max_bitrate_bps = 100000;
UpdateSendConfig(i, &send_config, &encoder_config,
&frame_generators[i]);
send_streams[i] = sender_call->CreateVideoSendStream(
send_config.Copy(), encoder_config.Copy());
send_streams[i]->Start();
VideoReceiveStream::Config receive_config(receiver_transport.get());
receive_config.rtp.remote_ssrc = ssrc;
receive_config.rtp.local_ssrc = test::CallTest::kReceiverLocalVideoSsrc;
VideoReceiveStream::Decoder decoder =
test::CreateMatchingDecoder(send_config.encoder_settings);
allocated_decoders.push_back(
std::unique_ptr<VideoDecoder>(decoder.decoder));
receive_config.decoders.push_back(decoder);
UpdateReceiveConfig(i, &receive_config);
receive_streams[i] =
receiver_call->CreateVideoReceiveStream(std::move(receive_config));
receive_streams[i]->Start();
frame_generators[i] = test::FrameGeneratorCapturer::Create(
width, height, 30, Clock::GetRealTimeClock());
send_streams[i]->SetSource(
frame_generators[i],
VideoSendStream::DegradationPreference::kMaintainFramerate);
frame_generators[i]->Start();
}
});
Wait();
task_queue_->SendTask([&]() {
for (size_t i = 0; i < kNumStreams; ++i) {
frame_generators[i]->Stop();
sender_call->DestroyVideoSendStream(send_streams[i]);
receiver_call->DestroyVideoReceiveStream(receive_streams[i]);
delete frame_generators[i];
}
sender_transport.reset();
receiver_transport.reset();
sender_call.reset();
receiver_call.reset();
});
}
protected:
virtual void Wait() = 0;
// Note: frame_generator is a point-to-pointer, since the actual instance
// hasn't been created at the time of this call. Only when packets/frames
// start flowing should this be dereferenced.
virtual void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) {}
virtual void UpdateReceiveConfig(size_t stream_index,
VideoReceiveStream::Config* receive_config) {
}
virtual test::DirectTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) {
return new test::DirectTransport(task_queue, sender_call,
payload_type_map_);
}
virtual test::DirectTransport* CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* receiver_call) {
return new test::DirectTransport(task_queue, receiver_call,
payload_type_map_);
}
test::SingleThreadedTaskQueueForTesting* const task_queue_;
};
// Each renderer verifies that it receives the expected resolution, and as soon
// as every renderer has received a frame, the test finishes.
TEST_P(EndToEndTest, SendsAndReceivesMultipleStreams) {
class VideoOutputObserver : public rtc::VideoSinkInterface<VideoFrame> {
public:
VideoOutputObserver(const MultiStreamTest::CodecSettings& settings,
uint32_t ssrc,
test::FrameGeneratorCapturer** frame_generator)
: settings_(settings),
ssrc_(ssrc),
frame_generator_(frame_generator),
done_(false, false) {}
void OnFrame(const VideoFrame& video_frame) override {
EXPECT_EQ(settings_.width, video_frame.width());
EXPECT_EQ(settings_.height, video_frame.height());
(*frame_generator_)->Stop();
done_.Set();
}
uint32_t Ssrc() { return ssrc_; }
bool Wait() { return done_.Wait(kDefaultTimeoutMs); }
private:
const MultiStreamTest::CodecSettings& settings_;
const uint32_t ssrc_;
test::FrameGeneratorCapturer** const frame_generator_;
rtc::Event done_;
};
class Tester : public MultiStreamTest {
public:
explicit Tester(test::SingleThreadedTaskQueueForTesting* task_queue)
: MultiStreamTest(task_queue) {}
virtual ~Tester() {}
protected:
void Wait() override {
for (const auto& observer : observers_) {
EXPECT_TRUE(observer->Wait()) << "Time out waiting for from on ssrc "
<< observer->Ssrc();
}
}
void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
observers_[stream_index].reset(new VideoOutputObserver(
codec_settings[stream_index], send_config->rtp.ssrcs.front(),
frame_generator));
}
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) override {
receive_config->renderer = observers_[stream_index].get();
}
private:
std::unique_ptr<VideoOutputObserver> observers_[kNumStreams];
} tester(&task_queue_);
tester.RunTest();
}
TEST_P(EndToEndTest, AssignsTransportSequenceNumbers) {
static const int kExtensionId = 5;
class RtpExtensionHeaderObserver : public test::DirectTransport {
public:
RtpExtensionHeaderObserver(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call,
const uint32_t& first_media_ssrc,
const std::map<uint32_t, uint32_t>& ssrc_map,
const std::map<uint8_t, MediaType>& payload_type_map)
: DirectTransport(task_queue, sender_call, payload_type_map),
done_(false, false),
parser_(RtpHeaderParser::Create()),
first_media_ssrc_(first_media_ssrc),
rtx_to_media_ssrcs_(ssrc_map),
padding_observed_(false),
rtx_padding_observed_(false),
retransmit_observed_(false),
started_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kExtensionId);
}
virtual ~RtpExtensionHeaderObserver() {}
bool SendRtp(const uint8_t* data,
size_t length,
const PacketOptions& options) override {
{
rtc::CritScope cs(&lock_);
if (IsDone())
return false;
if (started_) {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(data, length, &header));
bool drop_packet = false;
EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
EXPECT_EQ(options.packet_id,
header.extension.transportSequenceNumber);
if (!streams_observed_.empty()) {
// Unwrap packet id and verify uniqueness.
int64_t packet_id = unwrapper_.Unwrap(options.packet_id);
EXPECT_TRUE(received_packed_ids_.insert(packet_id).second);
}
// Drop (up to) every 17th packet, so we get retransmits.
// Only drop media, and not on the first stream (otherwise it will be
// hard to distinguish from padding, which is always sent on the first
// stream).
if (header.payloadType != kSendRtxPayloadType &&
header.ssrc != first_media_ssrc_ &&
header.extension.transportSequenceNumber % 17 == 0) {
dropped_seq_[header.ssrc].insert(header.sequenceNumber);
drop_packet = true;
}
if (header.payloadType == kSendRtxPayloadType) {
uint16_t original_sequence_number =
ByteReader<uint16_t>::ReadBigEndian(&data[header.headerLength]);
uint32_t original_ssrc =
rtx_to_media_ssrcs_.find(header.ssrc)->second;
std::set<uint16_t>* seq_no_map = &dropped_seq_[original_ssrc];
auto it = seq_no_map->find(original_sequence_number);
if (it != seq_no_map->end()) {
retransmit_observed_ = true;
seq_no_map->erase(it);
} else {
rtx_padding_observed_ = true;
}
} else {
streams_observed_.insert(header.ssrc);
}
if (IsDone())
done_.Set();
if (drop_packet)
return true;
}
}
return test::DirectTransport::SendRtp(data, length, options);
}
bool IsDone() {
bool observed_types_ok =
streams_observed_.size() == MultiStreamTest::kNumStreams &&
retransmit_observed_ && rtx_padding_observed_;
if (!observed_types_ok)
return false;
// We should not have any gaps in the sequence number range.
size_t seqno_range =
*received_packed_ids_.rbegin() - *received_packed_ids_.begin() + 1;
return seqno_range == received_packed_ids_.size();
}
bool Wait() {
{
// Can't be sure until this point that rtx_to_media_ssrcs_ etc have
// been initialized and are OK to read.
rtc::CritScope cs(&lock_);
started_ = true;
}
return done_.Wait(kDefaultTimeoutMs);
}
rtc::CriticalSection lock_;
rtc::Event done_;
std::unique_ptr<RtpHeaderParser> parser_;
SequenceNumberUnwrapper unwrapper_;
std::set<int64_t> received_packed_ids_;
std::set<uint32_t> streams_observed_;
std::map<uint32_t, std::set<uint16_t>> dropped_seq_;
const uint32_t& first_media_ssrc_;
const std::map<uint32_t, uint32_t>& rtx_to_media_ssrcs_;
bool padding_observed_;
bool rtx_padding_observed_;
bool retransmit_observed_;
bool started_;
};
class TransportSequenceNumberTester : public MultiStreamTest {
public:
explicit TransportSequenceNumberTester(
test::SingleThreadedTaskQueueForTesting* task_queue)
: MultiStreamTest(task_queue),
first_media_ssrc_(0),
observer_(nullptr) {}
virtual ~TransportSequenceNumberTester() {}
protected:
void Wait() override {
RTC_DCHECK(observer_);
EXPECT_TRUE(observer_->Wait());
}
void UpdateSendConfig(
size_t stream_index,
VideoSendStream::Config* send_config,
VideoEncoderConfig* encoder_config,
test::FrameGeneratorCapturer** frame_generator) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kExtensionId));
// Force some padding to be sent. Note that since we do send media
// packets we can not guarantee that a padding only packet is sent.
// Instead, padding will most likely be send as an RTX packet.
const int kPaddingBitrateBps = 50000;
encoder_config->max_bitrate_bps = 200000;
encoder_config->min_transmit_bitrate_bps =
encoder_config->max_bitrate_bps + kPaddingBitrateBps;
// Configure RTX for redundant payload padding.
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[stream_index]);
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
rtx_to_media_ssrcs_[kSendRtxSsrcs[stream_index]] =
send_config->rtp.ssrcs[0];
if (stream_index == 0)
first_media_ssrc_ = send_config->rtp.ssrcs[0];
}
void UpdateReceiveConfig(
size_t stream_index,
VideoReceiveStream::Config* receive_config) override {
receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
receive_config->rtp.extensions.clear();
receive_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kExtensionId));
receive_config->renderer = &fake_renderer_;
}
test::DirectTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
std::map<uint8_t, MediaType> payload_type_map =
MultiStreamTest::payload_type_map_;
RTC_DCHECK(payload_type_map.find(kSendRtxPayloadType) ==
payload_type_map.end());
payload_type_map[kSendRtxPayloadType] = MediaType::VIDEO;
observer_ = new RtpExtensionHeaderObserver(
task_queue, sender_call, first_media_ssrc_, rtx_to_media_ssrcs_,
payload_type_map);
return observer_;
}
private:
test::FakeVideoRenderer fake_renderer_;
uint32_t first_media_ssrc_;
std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
RtpExtensionHeaderObserver* observer_;
} tester(&task_queue_);
tester.RunTest();
}
class TransportFeedbackTester : public test::EndToEndTest {
public:
TransportFeedbackTester(bool feedback_enabled,
size_t num_video_streams,
size_t num_audio_streams)
: EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
feedback_enabled_(feedback_enabled),
num_video_streams_(num_video_streams),
num_audio_streams_(num_audio_streams),
receiver_call_(nullptr) {
// Only one stream of each supported for now.
EXPECT_LE(num_video_streams, 1u);
EXPECT_LE(num_audio_streams, 1u);
}
protected:
Action OnSendRtcp(const uint8_t* data, size_t length) override {
EXPECT_FALSE(HasTransportFeedback(data, length));
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
if (HasTransportFeedback(data, length))
observation_complete_.Set();
return SEND_PACKET;
}
bool HasTransportFeedback(const uint8_t* data, size_t length) const {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(data, length));
return parser.transport_feedback()->num_packets() > 0;
}
void PerformTest() override {
const int64_t kDisabledFeedbackTimeoutMs = 5000;
EXPECT_EQ(feedback_enabled_,
observation_complete_.Wait(feedback_enabled_
? test::CallTest::kDefaultTimeoutMs
: kDisabledFeedbackTimeoutMs));
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
receiver_call_ = receiver_call;
}
size_t GetNumVideoStreams() const override { return num_video_streams_; }
size_t GetNumAudioStreams() const override { return num_audio_streams_; }
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
(*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
}
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kTransportSequenceNumberUri, kExtensionId));
(*receive_configs)[0].rtp.extensions.clear();
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
(*receive_configs)[0].rtp.transport_cc = feedback_enabled_;
}
private:
static const int kExtensionId = 5;
const bool feedback_enabled_;
const size_t num_video_streams_;
const size_t num_audio_streams_;
Call* receiver_call_;
};
TEST_P(EndToEndTest, VideoReceivesTransportFeedback) {
TransportFeedbackTester test(true, 1, 0);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, VideoTransportFeedbackNotConfigured) {
TransportFeedbackTester test(false, 1, 0);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, AudioReceivesTransportFeedback) {
TransportFeedbackTester test(true, 0, 1);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, AudioTransportFeedbackNotConfigured) {
TransportFeedbackTester test(false, 0, 1);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, AudioVideoReceivesTransportFeedback) {
TransportFeedbackTester test(true, 1, 1);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, StopsSendingMediaWithoutFeedback) {
test::ScopedFieldTrials override_field_trials(
"WebRTC-CwndExperiment/Enabled-250/");
class TransportFeedbackTester : public test::EndToEndTest {
public:
TransportFeedbackTester(size_t num_video_streams, size_t num_audio_streams)
: EndToEndTest(::webrtc::EndToEndTest::kDefaultTimeoutMs),
num_video_streams_(num_video_streams),
num_audio_streams_(num_audio_streams),
media_sent_(0),
padding_sent_(0) {
// Only one stream of each supported for now.
EXPECT_LE(num_video_streams, 1u);
EXPECT_LE(num_audio_streams, 1u);
}
protected:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
const bool only_padding =
header.headerLength + header.paddingLength == length;
rtc::CritScope lock(&crit_);
if (only_padding) {
++padding_sent_;
} else {
++media_sent_;
EXPECT_LT(media_sent_, 40) << "Media sent without feedback.";
}
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* data, size_t length) override {
rtc::CritScope lock(&crit_);
if (media_sent_ > 20 && HasTransportFeedback(data, length)) {
return DROP_PACKET;
}
return SEND_PACKET;
}
bool HasTransportFeedback(const uint8_t* data, size_t length) const {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(data, length));
return parser.transport_feedback()->num_packets() > 0;
}
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.bitrate_config.max_bitrate_bps = 300000;
return config;
}
void PerformTest() override {
const int64_t kDisabledFeedbackTimeoutMs = 10000;
observation_complete_.Wait(kDisabledFeedbackTimeoutMs);
rtc::CritScope lock(&crit_);
EXPECT_GT(padding_sent_, 0);
}
size_t GetNumVideoStreams() const override { return num_video_streams_; }
size_t GetNumAudioStreams() const override { return num_audio_streams_; }
private:
const size_t num_video_streams_;
const size_t num_audio_streams_;
rtc::CriticalSection crit_;
int media_sent_ RTC_GUARDED_BY(crit_);
int padding_sent_ RTC_GUARDED_BY(crit_);
} test(1, 0);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ObserversEncodedFrames) {
class EncodedFrameTestObserver : public EncodedFrameObserver {
public:
EncodedFrameTestObserver()
: length_(0), frame_type_(kEmptyFrame), called_(false, false) {}
virtual ~EncodedFrameTestObserver() {}
virtual void EncodedFrameCallback(const EncodedFrame& encoded_frame) {
frame_type_ = encoded_frame.frame_type_;
length_ = encoded_frame.length_;
buffer_.reset(new uint8_t[length_]);
memcpy(buffer_.get(), encoded_frame.data_, length_);
called_.Set();
}
bool Wait() { return called_.Wait(kDefaultTimeoutMs); }
void ExpectEqualFrames(const EncodedFrameTestObserver& observer) {
ASSERT_EQ(length_, observer.length_)
<< "Observed frames are of different lengths.";
EXPECT_EQ(frame_type_, observer.frame_type_)
<< "Observed frames have different frame types.";
EXPECT_EQ(0, memcmp(buffer_.get(), observer.buffer_.get(), length_))
<< "Observed encoded frames have different content.";
}
private:
std::unique_ptr<uint8_t[]> buffer_;
size_t length_;
FrameType frame_type_;
rtc::Event called_;
};
EncodedFrameTestObserver post_encode_observer;
EncodedFrameTestObserver pre_decode_observer;
test::FrameForwarder forwarder;
std::unique_ptr<test::FrameGenerator> frame_generator;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([&]() {
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
video_send_config_.post_encode_callback = &post_encode_observer;
video_receive_configs_[0].pre_decode_callback = &pre_decode_observer;
CreateVideoStreams();
Start();
frame_generator = test::FrameGenerator::CreateSquareGenerator(
kDefaultWidth, kDefaultHeight);
video_send_stream_->SetSource(
&forwarder, VideoSendStream::DegradationPreference::kMaintainFramerate);
forwarder.IncomingCapturedFrame(*frame_generator->NextFrame());
});
EXPECT_TRUE(post_encode_observer.Wait())
<< "Timed out while waiting for send-side encoded-frame callback.";
EXPECT_TRUE(pre_decode_observer.Wait())
<< "Timed out while waiting for pre-decode encoded-frame callback.";
post_encode_observer.ExpectEqualFrames(pre_decode_observer);
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
TEST_P(EndToEndTest, ReceiveStreamSendsRemb) {
class RembObserver : public test::EndToEndTest {
public:
RembObserver() : EndToEndTest(kDefaultTimeoutMs) {}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = false;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
if (parser.remb()->num_packets() > 0) {
EXPECT_EQ(kReceiverLocalVideoSsrc, parser.remb()->sender_ssrc());
EXPECT_LT(0U, parser.remb()->bitrate_bps());
EXPECT_EQ(1U, parser.remb()->ssrcs().size());
EXPECT_EQ(kVideoSendSsrcs[0], parser.remb()->ssrcs()[0]);
observation_complete_.Set();
}
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for a "
"receiver RTCP REMB packet to be "
"sent.";
}
} test;
RunBaseTest(&test);
}
class BandwidthStatsTest : public test::EndToEndTest {
public:
explicit BandwidthStatsTest(bool send_side_bwe)
: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
sender_call_(nullptr),
receiver_call_(nullptr),
has_seen_pacer_delay_(false),
send_side_bwe_(send_side_bwe) {}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (!send_side_bwe_) {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kAbsSendTimeUri, test::kAbsSendTimeExtensionId));
(*receive_configs)[0].rtp.remb = true;
(*receive_configs)[0].rtp.transport_cc = false;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
Call::Stats sender_stats = sender_call_->GetStats();
Call::Stats receiver_stats = receiver_call_->GetStats();
if (!has_seen_pacer_delay_)
has_seen_pacer_delay_ = sender_stats.pacer_delay_ms > 0;
if (sender_stats.send_bandwidth_bps > 0 && has_seen_pacer_delay_) {
if (send_side_bwe_ || receiver_stats.recv_bandwidth_bps > 0)
observation_complete_.Set();
}
return SEND_PACKET;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
"non-zero bandwidth stats.";
}
private:
Call* sender_call_;
Call* receiver_call_;
bool has_seen_pacer_delay_;
const bool send_side_bwe_;
};
TEST_P(EndToEndTest, VerifySendSideBweStats) {
BandwidthStatsTest test(true);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, VerifyRecvSideBweStats) {
BandwidthStatsTest test(false);
RunBaseTest(&test);
}
// Verifies that it's possible to limit the send BWE by sending a REMB.
// This is verified by allowing the send BWE to ramp-up to >1000 kbps,
// then have the test generate a REMB of 500 kbps and verify that the send BWE
// is reduced to exactly 500 kbps. Then a REMB of 1000 kbps is generated and the
// test verifies that the send BWE ramps back up to exactly 1000 kbps.
TEST_P(EndToEndTest, RembWithSendSideBwe) {
class BweObserver : public test::EndToEndTest {
public:
BweObserver()
: EndToEndTest(kDefaultTimeoutMs),
sender_call_(nullptr),
clock_(Clock::GetRealTimeClock()),
sender_ssrc_(0),
remb_bitrate_bps_(1000000),
receive_transport_(nullptr),
stop_event_(false, false),
poller_thread_(&BitrateStatsPollingThread,
this,
"BitrateStatsPollingThread"),
state_(kWaitForFirstRampUp),
retransmission_rate_limiter_(clock_, 1000) {}
~BweObserver() {}
test::PacketTransport* CreateReceiveTransport(
test::SingleThreadedTaskQueueForTesting* task_queue) override {
receive_transport_ = new test::PacketTransport(
task_queue, nullptr, this, test::PacketTransport::kReceiver,
payload_type_map_, FakeNetworkPipe::Config());
return receive_transport_;
}
Call::Config GetSenderCallConfig() override {
Call::Config config(event_log_.get());
// Set a high start bitrate to reduce the test completion time.
config.bitrate_config.start_bitrate_bps = remb_bitrate_bps_;
return config;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
ASSERT_EQ(1u, send_config->rtp.ssrcs.size());
sender_ssrc_ = send_config->rtp.ssrcs[0];
encoder_config->max_bitrate_bps = 2000000;
ASSERT_EQ(1u, receive_configs->size());
RtpRtcp::Configuration config;
config.receiver_only = true;
config.clock = clock_;
config.outgoing_transport = receive_transport_;
config.retransmission_rate_limiter = &retransmission_rate_limiter_;
rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config));
rtp_rtcp_->SetRemoteSSRC((*receive_configs)[0].rtp.remote_ssrc);
rtp_rtcp_->SetSSRC((*receive_configs)[0].rtp.local_ssrc);
rtp_rtcp_->SetRTCPStatus(RtcpMode::kReducedSize);
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
}
static void BitrateStatsPollingThread(void* obj) {
static_cast<BweObserver*>(obj)->PollStats();
}
void PollStats() {
do {
if (sender_call_) {
Call::Stats stats = sender_call_->GetStats();
switch (state_) {
case kWaitForFirstRampUp:
if (stats.send_bandwidth_bps >= remb_bitrate_bps_) {
state_ = kWaitForRemb;
remb_bitrate_bps_ /= 2;
rtp_rtcp_->SetRemb(
remb_bitrate_bps_,
std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
rtp_rtcp_->SendRTCP(kRtcpRr);
}
break;
case kWaitForRemb:
if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
state_ = kWaitForSecondRampUp;
remb_bitrate_bps_ *= 2;
rtp_rtcp_->SetRemb(
remb_bitrate_bps_,
std::vector<uint32_t>(&sender_ssrc_, &sender_ssrc_ + 1));
rtp_rtcp_->SendRTCP(kRtcpRr);
}
break;
case kWaitForSecondRampUp:
if (stats.send_bandwidth_bps == remb_bitrate_bps_) {
observation_complete_.Set();
}
break;
}
}
} while (!stop_event_.Wait(1000));
}
void PerformTest() override {
poller_thread_.Start();
EXPECT_TRUE(Wait())
<< "Timed out while waiting for bitrate to change according to REMB.";
stop_event_.Set();
poller_thread_.Stop();
}
private:
enum TestState { kWaitForFirstRampUp, kWaitForRemb, kWaitForSecondRampUp };
Call* sender_call_;
Clock* const clock_;
uint32_t sender_ssrc_;
int remb_bitrate_bps_;
std::unique_ptr<RtpRtcp> rtp_rtcp_;
test::PacketTransport* receive_transport_;
rtc::Event stop_event_;
rtc::PlatformThread poller_thread_;
TestState state_;
RateLimiter retransmission_rate_limiter_;
} test;
RunBaseTest(&test);
}
TEST_P(EndToEndTest, StopSendingKeyframeRequestsForInactiveStream) {
class KeyframeRequestObserver : public test::EndToEndTest {
public:
explicit KeyframeRequestObserver(
test::SingleThreadedTaskQueueForTesting* task_queue)
: clock_(Clock::GetRealTimeClock()), task_queue_(task_queue) {}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
RTC_DCHECK_EQ(1, receive_streams.size());
send_stream_ = send_stream;
receive_stream_ = receive_streams[0];
}
void PerformTest() override {
bool frame_decoded = false;
int64_t start_time = clock_->TimeInMilliseconds();
while (clock_->TimeInMilliseconds() - start_time <= 5000) {
if (receive_stream_->GetStats().frames_decoded > 0) {
frame_decoded = true;
break;
}
SleepMs(100);
}
ASSERT_TRUE(frame_decoded);
task_queue_->SendTask([this]() { send_stream_->Stop(); });
SleepMs(10000);
ASSERT_EQ(
1U, receive_stream_->GetStats().rtcp_packet_type_counts.pli_packets);
}
private:
Clock* clock_;
VideoSendStream* send_stream_;
VideoReceiveStream* receive_stream_;
test::SingleThreadedTaskQueueForTesting* const task_queue_;
} test(&task_queue_);
RunBaseTest(&test);
}
class ProbingTest : public test::EndToEndTest {
public:
explicit ProbingTest(int start_bitrate_bps)
: clock_(Clock::GetRealTimeClock()),
start_bitrate_bps_(start_bitrate_bps),
state_(0),
sender_call_(nullptr) {}
~ProbingTest() {}
Call::Config GetSenderCallConfig() override {
Call::Config config(event_log_.get());
config.bitrate_config.start_bitrate_bps = start_bitrate_bps_;
return config;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
}
protected:
Clock* const clock_;
const int start_bitrate_bps_;
int state_;
Call* sender_call_;
};
// Flaky under MemorySanitizer: bugs.webrtc.org/7419
// Flaky on iOS bots: bugs.webrtc.org/7851
#if defined(MEMORY_SANITIZER)
TEST_P(EndToEndTest, DISABLED_InitialProbing) {
#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
TEST_P(EndToEndTest, DISABLED_InitialProbing) {
#else
TEST_P(EndToEndTest, InitialProbing) {
#endif
class InitialProbingTest : public ProbingTest {
public:
explicit InitialProbingTest(bool* success)
: ProbingTest(300000), success_(success) {
*success_ = false;
}
void PerformTest() override {
int64_t start_time_ms = clock_->TimeInMilliseconds();
do {
if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
break;
Call::Stats stats = sender_call_->GetStats();
// Initial probing is done with a x3 and x6 multiplier of the start
// bitrate, so a x4 multiplier is a high enough threshold.
if (stats.send_bandwidth_bps > 4 * 300000) {
*success_ = true;
break;
}
} while (!observation_complete_.Wait(20));
}
private:
const int kTimeoutMs = 1000;
bool* const success_;
};
bool success = false;
const int kMaxAttempts = 3;
for (int i = 0; i < kMaxAttempts; ++i) {
InitialProbingTest test(&success);
RunBaseTest(&test);
if (success)
return;
}
EXPECT_TRUE(success) << "Failed to perform mid initial probing ("
<< kMaxAttempts << " attempts).";
}
// Fails on Linux MSan: bugs.webrtc.org/7428
#if defined(MEMORY_SANITIZER)
TEST_P(EndToEndTest, DISABLED_TriggerMidCallProbing) {
// Fails on iOS bots: bugs.webrtc.org/7851
#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
TEST_P(EndToEndTest, DISABLED_TriggerMidCallProbing) {
#else
TEST_P(EndToEndTest, TriggerMidCallProbing) {
#endif
class TriggerMidCallProbingTest : public ProbingTest {
public:
TriggerMidCallProbingTest(
test::SingleThreadedTaskQueueForTesting* task_queue,
bool* success)
: ProbingTest(300000), success_(success), task_queue_(task_queue) {}
void PerformTest() override {
*success_ = false;
int64_t start_time_ms = clock_->TimeInMilliseconds();
do {
if (clock_->TimeInMilliseconds() - start_time_ms > kTimeoutMs)
break;
Call::Stats stats = sender_call_->GetStats();
switch (state_) {
case 0:
if (stats.send_bandwidth_bps > 5 * 300000) {
Call::Config::BitrateConfig bitrate_config;
bitrate_config.max_bitrate_bps = 100000;
task_queue_->SendTask([this, &bitrate_config]() {
sender_call_->SetBitrateConfig(bitrate_config);
});
++state_;
}
break;
case 1:
if (stats.send_bandwidth_bps < 110000) {
Call::Config::BitrateConfig bitrate_config;
bitrate_config.max_bitrate_bps = 2500000;
task_queue_->SendTask([this, &bitrate_config]() {
sender_call_->SetBitrateConfig(bitrate_config);
});
++state_;
}
break;
case 2:
// During high cpu load the pacer will not be able to pace packets
// at the correct speed, but if we go from 110 to 1250 kbps
// in 5 seconds then it is due to probing.
if (stats.send_bandwidth_bps > 1250000) {
*success_ = true;
observation_complete_.Set();
}
break;
}
} while (!observation_complete_.Wait(20));
}
private:
const int kTimeoutMs = 5000;
bool* const success_;
test::SingleThreadedTaskQueueForTesting* const task_queue_;
};
bool success = false;
const int kMaxAttempts = 3;
for (int i = 0; i < kMaxAttempts; ++i) {
TriggerMidCallProbingTest test(&task_queue_, &success);
RunBaseTest(&test);
if (success)
return;
}
EXPECT_TRUE(success) << "Failed to perform mid call probing (" << kMaxAttempts
<< " attempts).";
}
TEST_P(EndToEndTest, VerifyNackStats) {
static const int kPacketNumberToDrop = 200;
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(kLongTimeoutMs),
sent_rtp_packets_(0),
dropped_rtp_packet_(0),
dropped_rtp_packet_requested_(false),
send_stream_(nullptr),
start_runtime_ms_(-1) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
if (++sent_rtp_packets_ == kPacketNumberToDrop) {
std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
RTPHeader header;
EXPECT_TRUE(parser->Parse(packet, length, &header));
dropped_rtp_packet_ = header.sequenceNumber;
return DROP_PACKET;
}
VerifyStats();
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
const std::vector<uint16_t>& nacks = rtcp_parser.nack()->packet_ids();
if (!nacks.empty() && std::find(
nacks.begin(), nacks.end(), dropped_rtp_packet_) != nacks.end()) {
dropped_rtp_packet_requested_ = true;
}
return SEND_PACKET;
}
void VerifyStats() RTC_EXCLUSIVE_LOCKS_REQUIRED(&crit_) {
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 (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
stats.substreams.begin(); it != stats.substreams.end(); ++it) {
const VideoSendStream::StreamStats& stream_stats = it->second;
send_stream_nack_packets +=
stream_stats.rtcp_packet_type_counts.nack_packets;
}
for (size_t i = 0; i < receive_streams_.size(); ++i) {
VideoReceiveStream::Stats stats = receive_streams_[i]->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 = 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;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
(*receive_configs)[0].renderer = &fake_renderer_;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
receive_streams_ = receive_streams;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
}
test::FakeVideoRenderer fake_renderer_;
rtc::CriticalSection crit_;
uint64_t sent_rtp_packets_;
uint16_t dropped_rtp_packet_ RTC_GUARDED_BY(&crit_);
bool dropped_rtp_packet_requested_ RTC_GUARDED_BY(&crit_);
std::vector<VideoReceiveStream*> receive_streams_;
VideoSendStream* send_stream_;
int64_t start_runtime_ms_;
} test;
metrics::Reset();
RunBaseTest(&test);
EXPECT_EQ(
1, metrics::NumSamples("WebRTC.Video.UniqueNackRequestsSentInPercent"));
EXPECT_EQ(1, metrics::NumSamples(
"WebRTC.Video.UniqueNackRequestsReceivedInPercent"));
EXPECT_GT(metrics::MinSample("WebRTC.Video.NackPacketsSentPerMinute"), 0);
}
void EndToEndTest::VerifyHistogramStats(bool use_rtx,
bool use_fec,
bool screenshare) {
class StatsObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver(bool use_rtx, bool use_fec, bool screenshare)
: EndToEndTest(kLongTimeoutMs),
use_rtx_(use_rtx),
use_fec_(use_fec),
screenshare_(screenshare),
// This test uses NACK, so to send FEC we can't use a fake encoder.
vp8_encoder_(use_fec ? VP8Encoder::Create() : nullptr),
sender_call_(nullptr),
receiver_call_(nullptr),
start_runtime_ms_(-1),
num_frames_received_(0) {}
private:
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()) {
rtc::CritScope lock(&crit_);
++num_frames_received_;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (MinMetricRunTimePassed() && MinNumberOfFramesReceived())
observation_complete_.Set();
return SEND_PACKET;
}
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;
}
bool MinNumberOfFramesReceived() const {
const int kMinRequiredHistogramSamples = 200;
rtc::CritScope lock(&crit_);
return num_frames_received_ > kMinRequiredHistogramSamples;
}
void ModifyVideoConfigs(
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;
(*receive_configs)[0].renderer = this;
// FEC
if (use_fec_) {
send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType;
send_config->rtp.ulpfec.red_payload_type = kRedPayloadType;
send_config->encoder_settings.encoder = vp8_encoder_.get();
send_config->encoder_settings.payload_name = "VP8";
(*receive_configs)[0].decoders[0].payload_name = "VP8";
(*receive_configs)[0].rtp.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.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_ssrc = kSendRtxSsrcs[0];
(*receive_configs)[0]
.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
kFakeVideoSendPayloadType;
if (use_fec_) {
send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType;
(*receive_configs)[0]
.rtp.rtx_associated_payload_types[kRtxRedPayloadType] =
kSendRtxPayloadType;
}
}
// RTT needed for RemoteNtpTimeEstimator for the receive stream.
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = true;
encoder_config->content_type =
screenshare_ ? VideoEncoderConfig::ContentType::kScreen
: VideoEncoderConfig::ContentType::kRealtimeVideo;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
}
rtc::CriticalSection crit_;
const bool use_rtx_;
const bool use_fec_;
const bool screenshare_;
const std::unique_ptr<VideoEncoder> vp8_encoder_;
Call* sender_call_;
Call* receiver_call_;
int64_t start_runtime_ms_;
int num_frames_received_ RTC_GUARDED_BY(&crit_);
} test(use_rtx, use_fec, screenshare);
metrics::Reset();
RunBaseTest(&test);
std::string video_prefix =
screenshare ? "WebRTC.Video.Screenshare." : "WebRTC.Video.";
// The content type extension is disabled in non screenshare test,
// therefore no slicing on simulcast id should be present.
std::string video_suffix = screenshare ? ".S0" : "";
// Verify that stats have been updated once.
EXPECT_EQ(2, metrics::NumSamples("WebRTC.Call.LifetimeInSeconds"));
EXPECT_EQ(1, metrics::NumSamples(
"WebRTC.Call.TimeReceivingVideoRtpPacketsInSeconds"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.VideoBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.RtcpBitrateReceivedInBps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.BitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.EstimatedSendBitrateInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Call.PacerBitrateInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendStreamLifetimeInSeconds"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.ReceiveStreamLifetimeInSeconds"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.NackPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "NackPacketsReceivedPerMinute"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.FirPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "FirPacketsReceivedPerMinute"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.PliPacketsSentPerMinute"));
EXPECT_EQ(1,
metrics::NumSamples(video_prefix + "PliPacketsReceivedPerMinute"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "KeyFramesSentInPermille"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.KeyFramesReceivedInPermille"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentPacketsLostInPercent"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.ReceivedPacketsLostInPercent"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputWidthInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputHeightInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentWidthInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentHeightInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedWidthInPixels"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "ReceivedHeightInPixels"));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputWidthInPixels",
kDefaultWidth));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "InputHeightInPixels",
kDefaultHeight));
EXPECT_EQ(
1, metrics::NumEvents(video_prefix + "SentWidthInPixels", kDefaultWidth));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "SentHeightInPixels",
kDefaultHeight));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedWidthInPixels",
kDefaultWidth));
EXPECT_EQ(1, metrics::NumEvents(video_prefix + "ReceivedHeightInPixels",
kDefaultHeight));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InputFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SentFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodedFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderFramesPerSecond"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.JitterBufferDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.TargetDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.CurrentDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.OnewayDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EndToEndDelayMaxInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "InterframeDelayMaxInMs" +
video_suffix));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.RenderSqrtPixelsPerSecond"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "EncodeTimeInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.DecodeTimeInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "NumberOfPauseEvents"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PausedTimeInPercent"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "BitrateSentInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.BitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "MediaBitrateSentInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.MediaBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "PaddingBitrateSentInKbps"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.PaddingBitrateReceivedInKbps"));
EXPECT_EQ(
1, metrics::NumSamples(video_prefix + "RetransmittedBitrateSentInKbps"));
EXPECT_EQ(1, metrics::NumSamples(
"WebRTC.Video.RetransmittedBitrateReceivedInKbps"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.SendDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples(video_prefix + "SendSideDelayMaxInMs"));
int num_rtx_samples = use_rtx ? 1 : 0;
EXPECT_EQ(num_rtx_samples,
metrics::NumSamples("WebRTC.Video.RtxBitrateSentInKbps"));
EXPECT_EQ(num_rtx_samples,
metrics::NumSamples("WebRTC.Video.RtxBitrateReceivedInKbps"));
int num_red_samples = use_fec ? 1 : 0;
EXPECT_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.FecBitrateSentInKbps"));
EXPECT_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.FecBitrateReceivedInKbps"));
EXPECT_EQ(num_red_samples,
metrics::NumSamples("WebRTC.Video.ReceivedFecPacketsInPercent"));
}
#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_P(EndToEndTest, MAYBE_ContentTypeSwitches) {
class StatsObserver : public test::BaseTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver() : BaseTest(kLongTimeoutMs), 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()) {
rtc::CritScope lock(&crit_);
++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;
rtc::CritScope lock(&crit_);
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.
{
rtc::CritScope lock(&crit_);
num_frames_received_ = 0;
}
}
rtc::CriticalSection crit_;
int num_frames_received_ RTC_GUARDED_BY(&crit_);
} test;
metrics::Reset();
Call::Config send_config(test.GetSenderCallConfig());
Call::Config recv_config(test.GetReceiverCallConfig());
VideoEncoderConfig encoder_config_with_screenshare;
task_queue_.SendTask([this, &test, &send_config,
&recv_config, &encoder_config_with_screenshare]() {
CreateSenderCall(send_config);
CreateReceiverCall(recv_config);
receive_transport_.reset(test.CreateReceiveTransport(&task_queue_));
send_transport_.reset(
test.CreateSendTransport(&task_queue_, sender_call_.get()));
send_transport_->SetReceiver(receiver_call_->Receiver());
receive_transport_->SetReceiver(sender_call_->Receiver());
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
CreateSendConfig(1, 0, 0, send_transport_.get());
CreateMatchingReceiveConfigs(receive_transport_.get());
// Modify send and receive configs.
video_send_config_.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.
video_encoder_config_.content_type =
VideoEncoderConfig::ContentType::kRealtimeVideo;
// Second encoder config for the second part of the test uses screenshare
encoder_config_with_screenshare = video_encoder_config_.Copy();
encoder_config_with_screenshare.content_type =
VideoEncoderConfig::ContentType::kScreen;
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
test.PerformTest();
// Replace old send stream.
task_queue_.SendTask([this, &encoder_config_with_screenshare]() {
sender_call_->DestroyVideoSendStream(video_send_stream_);
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_.Copy(), encoder_config_with_screenshare.Copy());
video_send_stream_->SetSource(
frame_generator_capturer_.get(),
VideoSendStream::DegradationPreference::kBalanced);
video_send_stream_->Start();
});
// Continue to run test but now with screenshare.
test.PerformTest();
task_queue_.SendTask([this]() {
Stop();
DestroyStreams();
send_transport_.reset();
receive_transport_.reset();
DestroyCalls();
});
// Verify that stats have been updated for both screenshare and video.
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayInMs"));
EXPECT_EQ(1,
metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.EndToEndDelayMaxInMs"));
EXPECT_EQ(
1, metrics::NumSamples("WebRTC.Video.Screenshare.EndToEndDelayMaxInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayInMs"));
EXPECT_EQ(1,
metrics::NumSamples(
"WebRTC.Video.Screenshare.InterframeDelayInMs"));
EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.InterframeDelayMaxInMs"));
EXPECT_EQ(1,
metrics::NumSamples(
"WebRTC.Video.Screenshare.InterframeDelayMaxInMs"));
}
TEST_P(EndToEndTest, VerifyHistogramStatsWithRtx) {
const bool kEnabledRtx = true;
const bool kEnabledRed = false;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_P(EndToEndTest, VerifyHistogramStatsWithRed) {
const bool kEnabledRtx = false;
const bool kEnabledRed = true;
const bool kScreenshare = false;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
TEST_P(EndToEndTest, VerifyHistogramStatsWithScreenshare) {
const bool kEnabledRtx = false;
const bool kEnabledRed = false;
const bool kScreenshare = true;
VerifyHistogramStats(kEnabledRtx, kEnabledRed, kScreenshare);
}
void EndToEndTest::TestSendsSetSsrcs(size_t num_ssrcs,
bool send_single_ssrc_first) {
class SendsSetSsrcs : public test::EndToEndTest {
public:
SendsSetSsrcs(const uint32_t* ssrcs,
size_t num_ssrcs,
bool send_single_ssrc_first)
: EndToEndTest(kDefaultTimeoutMs),
num_ssrcs_(num_ssrcs),
send_single_ssrc_first_(send_single_ssrc_first),
ssrcs_to_observe_(num_ssrcs),
expect_single_ssrc_(send_single_ssrc_first),
send_stream_(nullptr) {
for (size_t i = 0; i < num_ssrcs; ++i)
valid_ssrcs_[ssrcs[i]] = true;
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
EXPECT_TRUE(valid_ssrcs_[header.ssrc])
<< "Received unknown SSRC: " << header.ssrc;
if (!valid_ssrcs_[header.ssrc])
observation_complete_.Set();
if (!is_observed_[header.ssrc]) {
is_observed_[header.ssrc] = true;
--ssrcs_to_observe_;
if (expect_single_ssrc_) {
expect_single_ssrc_ = false;
observation_complete_.Set();
}
}
if (ssrcs_to_observe_ == 0)
observation_complete_.Set();
return SEND_PACKET;
}
size_t GetNumVideoStreams() const override { return num_ssrcs_; }
// This test use other VideoStream settings than the the default settings
// implemented in DefaultVideoStreamFactory. Therefore this test implement
// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
streams[i].min_bitrate_bps = 10000;
streams[i].target_bitrate_bps = 15000;
streams[i].max_bitrate_bps = 20000;
}
return streams;
}
};
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
encoder_config->video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>();
video_encoder_config_all_streams_ = encoder_config->Copy();
if (send_single_ssrc_first_)
encoder_config->number_of_streams = 1;
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for "
<< (send_single_ssrc_first_ ? "first SSRC."
: "SSRCs.");
if (send_single_ssrc_first_) {
// Set full simulcast and continue with the rest of the SSRCs.
send_stream_->ReconfigureVideoEncoder(
std::move(video_encoder_config_all_streams_));
EXPECT_TRUE(Wait()) << "Timed out while waiting on additional SSRCs.";
}
}
private:
std::map<uint32_t, bool> valid_ssrcs_;
std::map<uint32_t, bool> is_observed_;
const size_t num_ssrcs_;
const bool send_single_ssrc_first_;
size_t ssrcs_to_observe_;
bool expect_single_ssrc_;
VideoSendStream* send_stream_;
VideoEncoderConfig video_encoder_config_all_streams_;
} test(kVideoSendSsrcs, num_ssrcs, send_single_ssrc_first);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, ReportsSetEncoderRates) {
class EncoderRateStatsTest : public test::EndToEndTest,
public test::FakeEncoder {
public:
explicit EncoderRateStatsTest(
test::SingleThreadedTaskQueueForTesting* task_queue)
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
task_queue_(task_queue),
send_stream_(nullptr),
bitrate_kbps_(0) {}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
RTC_DCHECK_EQ(1, encoder_config->number_of_streams);
}
int32_t SetRateAllocation(const BitrateAllocation& rate_allocation,
uint32_t framerate) override {
// Make sure not to trigger on any default zero bitrates.
if (rate_allocation.get_sum_bps() == 0)
return 0;
rtc::CritScope lock(&crit_);
bitrate_kbps_ = rate_allocation.get_sum_kbps();
observation_complete_.Set();
return 0;
}
void PerformTest() override {
ASSERT_TRUE(Wait())
<< "Timed out while waiting for encoder SetRates() call.";
task_queue_->SendTask([this]() {
WaitForEncoderTargetBitrateMatchStats();
send_stream_->Stop();
WaitForStatsReportZeroTargetBitrate();
send_stream_->Start();
WaitForEncoderTargetBitrateMatchStats();
});
}
void WaitForEncoderTargetBitrateMatchStats() {
for (int i = 0; i < kDefaultTimeoutMs; ++i) {
VideoSendStream::Stats stats = send_stream_->GetStats();
{
rtc::CritScope lock(&crit_);
if ((stats.target_media_bitrate_bps + 500) / 1000 ==
static_cast<int>(bitrate_kbps_)) {
return;
}
}
SleepMs(1);
}
FAIL()
<< "Timed out waiting for stats reporting the currently set bitrate.";
}
void WaitForStatsReportZeroTargetBitrate() {
for (int i = 0; i < kDefaultTimeoutMs; ++i) {
if (send_stream_->GetStats().target_media_bitrate_bps == 0) {
return;
}
SleepMs(1);
}
FAIL() << "Timed out waiting for stats reporting zero bitrate.";
}
private:
test::SingleThreadedTaskQueueForTesting* const task_queue_;
rtc::CriticalSection crit_;
VideoSendStream* send_stream_;
uint32_t bitrate_kbps_ RTC_GUARDED_BY(crit_);
} test(&task_queue_);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, GetStats) {
static const int kStartBitrateBps = 3000000;
static const int kExpectedRenderDelayMs = 20;
class ReceiveStreamRenderer : public rtc::VideoSinkInterface<VideoFrame> {
public:
ReceiveStreamRenderer() {}
private:
void OnFrame(const VideoFrame& video_frame) override {}
};
class StatsObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver()
: EndToEndTest(kLongTimeoutMs),
encoder_(Clock::GetRealTimeClock(), 10),
send_stream_(nullptr),
expected_send_ssrcs_(),
check_stats_event_(false, false) {}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
// Drop every 25th packet => 4% loss.
static const int kPacketLossFrac = 25;
RTPHeader header;
RtpUtility::RtpHeaderParser parser(packet, length);
if (parser.Parse(&header) &&
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;
}
void OnFrame(const VideoFrame& video_frame) override {
// Ensure that we have at least 5ms send side delay.
SleepMs(5);
}
bool CheckReceiveStats() {
for (size_t i = 0; i < receive_streams_.size(); ++i) {
VideoReceiveStream::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_["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.rtcp_stats.packets_lost != 0 ||
stats.rtcp_stats.extended_highest_sequence_number != 0 ||
stats.rtcp_stats.fraction_lost != 0 || stats.rtcp_stats.jitter != 0;
receive_stats_filled_["DataCountersUpdated"] |=
stats.rtp_stats.transmitted.payload_bytes != 0 ||
stats.rtp_stats.fec.packets != 0 ||
stats.rtp_stats.transmitted.header_bytes != 0 ||
stats.rtp_stats.transmitted.packets != 0 ||
stats.rtp_stats.transmitted.padding_bytes != 0 ||
stats.rtp_stats.retransmitted.packets != 0;
receive_stats_filled_["CodecStats"] |=
stats.target_delay_ms != 0 || stats.discarded_packets != 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;
assert(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 = 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;
send_stats_filled_["EncoderImplementationName"] |=
stats.encoder_implementation_name ==
test::FakeEncoder::kImplementationName;
send_stats_filled_["EncoderPreferredBitrate"] |=
stats.preferred_media_bitrate_bps > 0;
for (std::map<uint32_t, VideoSendStream::StreamStats>::const_iterator it =
stats.substreams.begin();
it != stats.substreams.end(); ++it) {
if (expected_send_ssrcs_.find(it->first) == expected_send_ssrcs_.end())
continue; // Probably RTX.
send_stats_filled_[CompoundKey("CapturedFrameRate", it->first)] |=
stats.input_frame_rate != 0;
const VideoSendStream::StreamStats& stream_stats = it->second;
send_stats_filled_[CompoundKey("StatisticsUpdated", it->first)] |=
stream_stats.rtcp_stats.packets_lost != 0 ||
stream_stats.rtcp_stats.extended_highest_sequence_number != 0 ||
stream_stats.rtcp_stats.fraction_lost != 0;
send_stats_filled_[CompoundKey("DataCountersUpdated", it->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",
it->first)] |=
stream_stats.total_bitrate_bps != 0;
send_stats_filled_[CompoundKey("BitrateStatisticsObserver.Retransmit",
it->first)] |=
stream_stats.retransmit_bitrate_bps != 0;
send_stats_filled_[CompoundKey("FrameCountObserver", it->first)] |=
stream_stats.frame_counts.delta_frames != 0 ||
stream_stats.frame_counts.key_frames != 0;
send_stats_filled_[CompoundKey("OutgoingRate", it->first)] |=
stats.encode_frame_rate != 0;
send_stats_filled_[CompoundKey("Delay", it->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) {
std::ostringstream oss;
oss << name << "_" << ssrc;
return oss.str();
}
bool AllStatsFilled(const std::map<std::string, bool>& stats_map) {
for (const auto& stat : stats_map) {
if (!stat.second)
return false;
}
return true;
}
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
FakeNetworkPipe::Config network_config;
network_config.loss_percent = 5;
return new test::PacketTransport(task_queue, sender_call, this,
test::PacketTransport::kSender,
payload_type_map_, network_config);
}
Call::Config GetSenderCallConfig() override {
Call::Config config = EndToEndTest::GetSenderCallConfig();
config.bitrate_config.start_bitrate_bps = kStartBitrateBps;
return config;
}
// This test use other VideoStream settings than the the default settings
// implemented in DefaultVideoStreamFactory. Therefore this test implement
// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
streams[i].min_bitrate_bps = 10000;
streams[i].target_bitrate_bps = 15000;
streams[i].max_bitrate_bps = 20000;
}
return streams;
}
};
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
encoder_config->video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>();
send_config->pre_encode_callback = this; // Used to inject delay.
expected_cname_ = 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].renderer = &receive_stream_renderer_;
(*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 = &encoder_;
}
size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
send_stream_ = send_stream;
receive_streams_ = receive_streams;
}
void PerformTest() override {
Clock* clock = Clock::GetRealTimeClock();
int64_t now = clock->TimeInMilliseconds();
int64_t stop_time = now + test::CallTest::kLongTimeoutMs;
bool receive_ok = false;
bool send_ok = false;
while (now < stop_time) {
if (!receive_ok)
receive_ok = CheckReceiveStats();
if (!send_ok)
send_ok = CheckSendStats();
if (receive_ok && send_ok)
return;
int64_t time_until_timout_ = stop_time - now;
if (time_until_timout_ > 0)
check_stats_event_.Wait(time_until_timout_);
now = clock->TimeInMilliseconds();
}
ADD_FAILURE() << "Timed out waiting for filled stats.";
for (std::map<std::string, bool>::const_iterator it =
receive_stats_filled_.begin();
it != receive_stats_filled_.end(); ++it) {
if (!it->second) {
ADD_FAILURE() << "Missing receive stats: " << it->first;
}
}
for (std::map<std::string, bool>::const_iterator it =
send_stats_filled_.begin();
it != send_stats_filled_.end(); ++it) {
if (!it->second) {
ADD_FAILURE() << "Missing send stats: " << it->first;
}
}
}
test::DelayedEncoder encoder_;
std::vector<VideoReceiveStream*> receive_streams_;
std::map<std::string, bool> receive_stats_filled_;
VideoSendStream* send_stream_;
std::map<std::string, bool> send_stats_filled_;
std::vector<uint32_t> expected_receive_ssrcs_;
std::set<uint32_t> expected_send_ssrcs_;
std::string expected_cname_;
rtc::Event check_stats_event_;
ReceiveStreamRenderer receive_stream_renderer_;
} test;
RunBaseTest(&test);
}
TEST_P(EndToEndTest, TimingFramesAreReported) {
static const int kExtensionId = 5;
class StatsObserver : public test::EndToEndTest {
public:
StatsObserver() : EndToEndTest(kLongTimeoutMs) {}
private:
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(
RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
for (size_t i = 0; i < receive_configs->size(); ++i) {
(*receive_configs)[i].rtp.extensions.clear();
(*receive_configs)[i].rtp.extensions.push_back(
RtpExtension(RtpExtension::kVideoTimingUri, kExtensionId));
}
}
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
receive_streams_ = receive_streams;
}
void PerformTest() override {
// No frames reported initially.
for (size_t i = 0; i < receive_streams_.size(); ++i) {
EXPECT_FALSE(receive_streams_[i]->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.
for (size_t i = 0; i < receive_streams_.size(); ++i) {
EXPECT_TRUE(receive_streams_[i]->GetStats().timing_frame_info);
}
}
std::vector<VideoReceiveStream*> receive_streams_;
} test;
RunBaseTest(&test);
}
class RtcpXrObserver : public test::EndToEndTest {
public:
RtcpXrObserver(bool enable_rrtr, bool enable_target_bitrate,
bool enable_zero_target_bitrate)
: EndToEndTest(test::CallTest::kDefaultTimeoutMs),
enable_rrtr_(enable_rrtr),
enable_target_bitrate_(enable_target_bitrate),
enable_zero_target_bitrate_(enable_zero_target_bitrate),
sent_rtcp_sr_(0),
sent_rtcp_rr_(0),
sent_rtcp_rrtr_(0),
sent_rtcp_target_bitrate_(false),
sent_zero_rtcp_target_bitrate_(false),
sent_rtcp_dlrr_(0) {}
private:
// Receive stream should send RR packets (and RRTR packets if enabled).
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
sent_rtcp_rr_ += parser.receiver_report()->num_packets();
EXPECT_EQ(0, parser.sender_report()->num_packets());
EXPECT_GE(1, parser.xr()->num_packets());
if (parser.xr()->num_packets() > 0) {
if (parser.xr()->rrtr())
++sent_rtcp_rrtr_;
EXPECT_FALSE(parser.xr()->dlrr());
}
return SEND_PACKET;
}
// Send stream should send SR packets (and DLRR packets if enabled).
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet, length));
sent_rtcp_sr_ += parser.sender_report()->num_packets();
EXPECT_LE(parser.xr()->num_packets(), 1);
if (parser.xr()->num_packets() > 0) {
EXPECT_FALSE(parser.xr()->rrtr());
if (parser.xr()->dlrr())
++sent_rtcp_dlrr_;
if (parser.xr()->target_bitrate()) {
sent_rtcp_target_bitrate_ = true;
for (const rtcp::TargetBitrate::BitrateItem& item :
parser.xr()->target_bitrate()->GetTargetBitrates()) {
if (item.target_bitrate_kbps == 0) {
sent_zero_rtcp_target_bitrate_ = true;
break;
}
}
}
}
if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve &&
sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve &&
(sent_rtcp_target_bitrate_ || !enable_target_bitrate_) &&
(sent_zero_rtcp_target_bitrate_ || !enable_zero_target_bitrate_)) {
if (enable_rrtr_) {
EXPECT_GT(sent_rtcp_rrtr_, 0);
EXPECT_GT(sent_rtcp_dlrr_, 0);
} else {
EXPECT_EQ(sent_rtcp_rrtr_, 0);
EXPECT_EQ(sent_rtcp_dlrr_, 0);
}
EXPECT_EQ(enable_target_bitrate_, sent_rtcp_target_bitrate_);
EXPECT_EQ(enable_zero_target_bitrate_, sent_zero_rtcp_target_bitrate_);
observation_complete_.Set();
}
return SEND_PACKET;
}
size_t GetNumVideoStreams() const override {
// When sending a zero target bitrate, we use two spatial layers so that
// we'll still have a layer with non-zero bitrate.
return enable_zero_target_bitrate_ ? 2 : 1;
}
// This test uses VideoStream settings different from the the default one
// implemented in DefaultVideoStreamFactory, so it implements its own
// VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class ZeroTargetVideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
ZeroTargetVideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
// Set one of the streams' target bitrates to zero to test that a
// bitrate of 0 can be signalled.
streams[encoder_config.number_of_streams-1].min_bitrate_bps = 0;
streams[encoder_config.number_of_streams-1].target_bitrate_bps = 0;
streams[encoder_config.number_of_streams-1].max_bitrate_bps = 0;
return streams;
}
};
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
if (enable_zero_target_bitrate_) {
encoder_config->video_stream_factory =
new rtc::RefCountedObject<ZeroTargetVideoStreamFactory>();
// Configure VP8 to be able to use simulcast.
send_config->encoder_settings.payload_name = "VP8";
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->encoder_settings.payload_type;
(*receive_configs)[0].decoders[0].payload_name =
send_config->encoder_settings.payload_name;
}
if (enable_target_bitrate_) {
// TargetBitrate only signaled for screensharing.
encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen;
}
(*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize;
(*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report =
enable_rrtr_;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for RTCP SR/RR packets to be sent.";
}
static const int kNumRtcpReportPacketsToObserve = 5;
rtc::CriticalSection crit_;
const bool enable_rrtr_;
const bool enable_target_bitrate_;
const bool enable_zero_target_bitrate_;
int sent_rtcp_sr_;
int sent_rtcp_rr_ RTC_GUARDED_BY(&crit_);
int sent_rtcp_rrtr_ RTC_GUARDED_BY(&crit_);
bool sent_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
bool sent_zero_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_);
int sent_rtcp_dlrr_;
};
TEST_P(EndToEndTest, TestExtendedReportsWithRrtrWithoutTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/false,
/*enable_zero_target_bitrate=*/false);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtrWithoutTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/false,
/*enable_zero_target_bitrate=*/false);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, TestExtendedReportsWithRrtrWithTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/false);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, TestExtendedReportsWithoutRrtrWithTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/false);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, TestExtendedReportsCanSignalZeroTargetBitrate) {
RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true,
/*enable_zero_target_bitrate=*/true);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, TestReceivedRtpPacketStats) {
static const size_t kNumRtpPacketsToSend = 5;
class ReceivedRtpStatsObserver : public test::EndToEndTest {
public:
ReceivedRtpStatsObserver()
: EndToEndTest(kDefaultTimeoutMs),
receive_stream_(nullptr),
sent_rtp_(0) {}
private:
void OnVideoStreamsCreated(
VideoSendStream* send_stream,
const std::vector<VideoReceiveStream*>& receive_streams) override {
receive_stream_ = receive_streams[0];
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
if (sent_rtp_ >= kNumRtpPacketsToSend) {
VideoReceiveStream::Stats stats = receive_stream_->GetStats();
if (kNumRtpPacketsToSend == stats.rtp_stats.transmitted.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.";
}
VideoReceiveStream* receive_stream_;
uint32_t sent_rtp_;
} test;
RunBaseTest(&test);
}
TEST_P(EndToEndTest, SendsSetSsrc) {
TestSendsSetSsrcs(1, false);
}
TEST_P(EndToEndTest, SendsSetSimulcastSsrcs) {
TestSendsSetSsrcs(kNumSimulcastStreams, false);
}
TEST_P(EndToEndTest, CanSwitchToUseAllSsrcs) {
TestSendsSetSsrcs(kNumSimulcastStreams, true);
}
TEST_P(EndToEndTest, DISABLED_RedundantPayloadsTransmittedOnAllSsrcs) {
class ObserveRedundantPayloads: public test::EndToEndTest {
public:
ObserveRedundantPayloads()
: EndToEndTest(kDefaultTimeoutMs),
ssrcs_to_observe_(kNumSimulcastStreams) {
for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
registered_rtx_ssrc_[kSendRtxSsrcs[i]] = true;
}
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
if (!registered_rtx_ssrc_[header.ssrc])
return SEND_PACKET;
EXPECT_LE(header.headerLength + header.paddingLength, length);
const bool packet_is_redundant_payload =
header.headerLength + header.paddingLength < length;
if (!packet_is_redundant_payload)
return SEND_PACKET;
if (!observed_redundant_retransmission_[header.ssrc]) {
observed_redundant_retransmission_[header.ssrc] = true;
if (--ssrcs_to_observe_ == 0)
observation_complete_.Set();
}
return SEND_PACKET;
}
size_t GetNumVideoStreams() const override { return kNumSimulcastStreams; }
// This test use other VideoStream settings than the the default settings
// implemented in DefaultVideoStreamFactory. Therefore this test implement
// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
streams[i].min_bitrate_bps = 10000;
streams[i].target_bitrate_bps = 15000;
streams[i].max_bitrate_bps = 20000;
}
return streams;
}
};
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
// Set low simulcast bitrates to not have to wait for bandwidth ramp-up.
encoder_config->video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>();
send_config->rtp.rtx.payload_type = kSendRtxPayloadType;
for (size_t i = 0; i < kNumSimulcastStreams; ++i)
send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
// Significantly higher than max bitrates for all video streams -> forcing
// padding to trigger redundant padding on all RTX SSRCs.
encoder_config->min_transmit_bitrate_bps = 100000;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out while waiting for redundant payloads on all SSRCs.";
}
private:
size_t ssrcs_to_observe_;
std::map<uint32_t, bool> observed_redundant_retransmission_;
std::map<uint32_t, bool> registered_rtx_ssrc_;
} test;
RunBaseTest(&test);
}
void EndToEndTest::TestRtpStatePreservation(bool use_rtx,
bool provoke_rtcpsr_before_rtp) {
// This test uses other VideoStream settings than the the default settings
// implemented in DefaultVideoStreamFactory. Therefore this test implements
// its own VideoEncoderConfig::VideoStreamFactoryInterface which is created
// in ModifyVideoConfigs.
class VideoStreamFactory
: public VideoEncoderConfig::VideoStreamFactoryInterface {
public:
VideoStreamFactory() {}
private:
std::vector<VideoStream> CreateEncoderStreams(
int width,
int height,
const VideoEncoderConfig& encoder_config) override {
std::vector<VideoStream> streams =
test::CreateVideoStreams(width, height, encoder_config);
if (encoder_config.number_of_streams > 1) {
// Lower bitrates so that all streams send initially.
RTC_DCHECK_EQ(3, encoder_config.number_of_streams);
for (size_t i = 0; i < encoder_config.number_of_streams; ++i) {
streams[i].min_bitrate_bps = 10000;
streams[i].target_bitrate_bps = 15000;
streams[i].max_bitrate_bps = 20000;
}
} else {
// Use the same total bitrates when sending a single stream to avoid
// lowering
// the bitrate estimate and requiring a subsequent rampup.
streams[0].min_bitrate_bps = 3 * 10000;
streams[0].target_bitrate_bps = 3 * 15000;
streams[0].max_bitrate_bps = 3 * 20000;
}
return streams;
}
};
class RtpSequenceObserver : public test::RtpRtcpObserver {
public:
explicit RtpSequenceObserver(bool use_rtx)
: test::RtpRtcpObserver(kDefaultTimeoutMs),
ssrcs_to_observe_(kNumSimulcastStreams) {
for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
ssrc_is_rtx_[kVideoSendSsrcs[i]] = false;
if (use_rtx)
ssrc_is_rtx_[kSendRtxSsrcs[i]] = true;
}
}
void ResetExpectedSsrcs(size_t num_expected_ssrcs) {
rtc::CritScope lock(&crit_);
ssrc_observed_.clear();
ssrcs_to_observe_ = num_expected_ssrcs;
}
private:
void ValidateTimestampGap(uint32_t ssrc,
uint32_t timestamp,
bool only_padding)
RTC_EXCLUSIVE_LOCKS_REQUIRED(crit_) {
static const int32_t kMaxTimestampGap = kDefaultTimeoutMs * 90;
auto timestamp_it = last_observed_timestamp_.find(ssrc);
if (timestamp_it == last_observed_timestamp_.end()) {
EXPECT_FALSE(only_padding);
last_observed_timestamp_[ssrc] = timestamp;
} else {
// Verify timestamps are reasonably close.
uint32_t latest_observed = timestamp_it->second;
// Wraparound handling is unnecessary here as long as an int variable
// is used to store the result.
int32_t timestamp_gap = timestamp - latest_observed;
EXPECT_LE(std::abs(timestamp_gap), kMaxTimestampGap)
<< "Gap in timestamps (" << latest_observed << " -> " << timestamp
<< ") too large for SSRC: " << ssrc << ".";
timestamp_it->second = timestamp;
}
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
const uint32_t ssrc = header.ssrc;
const int64_t sequence_number =
seq_numbers_unwrapper_.Unwrap(header.sequenceNumber);
const uint32_t timestamp = header.timestamp;
const bool only_padding =
header.headerLength + header.paddingLength == length;
EXPECT_TRUE(ssrc_is_rtx_.find(ssrc) != ssrc_is_rtx_.end())
<< "Received SSRC that wasn't configured: " << ssrc;
static const int64_t kMaxSequenceNumberGap = 100;
std::list<int64_t>* seq_numbers = &last_observed_seq_numbers_[ssrc];
if (seq_numbers->empty()) {
seq_numbers->push_back(sequence_number);
} else {
// We shouldn't get replays of previous sequence numbers.
for (int64_t observed : *seq_numbers) {
EXPECT_NE(observed, sequence_number)
<< "Received sequence number " << sequence_number
<< " for SSRC " << ssrc << " 2nd time.";
}
// Verify sequence numbers are reasonably close.
int64_t latest_observed = seq_numbers->back();
int64_t sequence_number_gap = sequence_number - latest_observed;
EXPECT_LE(std::abs(sequence_number_gap), kMaxSequenceNumberGap)
<< "Gap in sequence numbers (" << latest_observed << " -> "
<< sequence_number << ") too large for SSRC: " << ssrc << ".";
seq_numbers->push_back(sequence_number);
if (seq_numbers->size() >= kMaxSequenceNumberGap) {
seq_numbers->pop_front();
}
}
if (!ssrc_is_rtx_[ssrc]) {
rtc::CritScope lock(&crit_);
ValidateTimestampGap(ssrc, timestamp, only_padding);
// Wait for media packets on all ssrcs.
if (!ssrc_observed_[ssrc] && !only_padding) {
ssrc_observed_[ssrc] = true;
if (--ssrcs_to_observe_ == 0)
observation_complete_.Set();
}
}
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
test::RtcpPacketParser rtcp_parser;
rtcp_parser.Parse(packet, length);
if (rtcp_parser.sender_report()->num_packets() > 0) {
uint32_t ssrc = rtcp_parser.sender_report()->sender_ssrc();
uint32_t rtcp_timestamp = rtcp_parser.sender_report()->rtp_timestamp();
rtc::CritScope lock(&crit_);
ValidateTimestampGap(ssrc, rtcp_timestamp, false);
}
return SEND_PACKET;
}
SequenceNumberUnwrapper seq_numbers_unwrapper_;
std::map<uint32_t, std::list<int64_t>> last_observed_seq_numbers_;
std::map<uint32_t, uint32_t> last_observed_timestamp_;
std::map<uint32_t, bool> ssrc_is_rtx_;
rtc::CriticalSection crit_;
size_t ssrcs_to_observe_ RTC_GUARDED_BY(crit_);
std::map<uint32_t, bool> ssrc_observed_ RTC_GUARDED_BY(crit_);
} observer(use_rtx);
std::unique_ptr<test::PacketTransport> send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
Call::Config config(event_log_.get());
VideoEncoderConfig one_stream;
task_queue_.SendTask([this, &observer, &send_transport, &receive_transport,
&config, &one_stream, use_rtx]() {
CreateCalls(config, config);
send_transport = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, sender_call_.get(), &observer,
test::PacketTransport::kSender, payload_type_map_,
FakeNetworkPipe::Config());
receive_transport = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
payload_type_map_, FakeNetworkPipe::Config());
send_transport->SetReceiver(receiver_call_->Receiver());
receive_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(kNumSimulcastStreams, 0, 0, send_transport.get());
if (use_rtx) {
for (size_t i = 0; i < kNumSimulcastStreams; ++i) {
video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[i]);
}
video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
}
video_encoder_config_.video_stream_factory =
new rtc::RefCountedObject<VideoStreamFactory>();
// Use the same total bitrates when sending a single stream to avoid
// lowering the bitrate estimate and requiring a subsequent rampup.
one_stream = video_encoder_config_.Copy();
// one_stream.streams.resize(1);
one_stream.number_of_streams = 1;
CreateMatchingReceiveConfigs(receive_transport.get());
CreateVideoStreams();
CreateFrameGeneratorCapturer(30, 1280, 720);
Start();
});
EXPECT_TRUE(observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
// Test stream resetting more than once to make sure that the state doesn't
// get set once (this could be due to using std::map::insert for instance).
for (size_t i = 0; i < 3; ++i) {
task_queue_.SendTask([&]() {
frame_generator_capturer_->Stop();
sender_call_->DestroyVideoSendStream(video_send_stream_);
// Re-create VideoSendStream with only one stream.
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_.Copy(), one_stream.Copy());
video_send_stream_->Start();
if (provoke_rtcpsr_before_rtp) {
// Rapid Resync Request forces sending RTCP Sender Report back.
// Using this request speeds up this test because then there is no need
// to wait for a second for periodic Sender Report.
rtcp::RapidResyncRequest force_send_sr_back_request;
rtc::Buffer packet = force_send_sr_back_request.Build();
static_cast<webrtc::test::DirectTransport*>(receive_transport.get())
->SendRtcp(packet.data(), packet.size());
}
CreateFrameGeneratorCapturer(30, 1280, 720);
frame_generator_capturer_->Start();
});
observer.ResetExpectedSsrcs(1);
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
// Reconfigure back to use all streams.
task_queue_.SendTask([this]() {
video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
});
observer.ResetExpectedSsrcs(kNumSimulcastStreams);
EXPECT_TRUE(observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
// Reconfigure down to one stream.
task_queue_.SendTask([this, &one_stream]() {
video_send_stream_->ReconfigureVideoEncoder(one_stream.Copy());
});
observer.ResetExpectedSsrcs(1);
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for single RTP packet.";
// Reconfigure back to use all streams.
task_queue_.SendTask([this]() {
video_send_stream_->ReconfigureVideoEncoder(video_encoder_config_.Copy());
});
observer.ResetExpectedSsrcs(kNumSimulcastStreams);
EXPECT_TRUE(observer.Wait())
<< "Timed out waiting for all SSRCs to send packets.";
}
task_queue_.SendTask([this, &send_transport, &receive_transport]() {
Stop();
DestroyStreams();
send_transport.reset();
receive_transport.reset();
DestroyCalls();
});
}
TEST_P(EndToEndTest, RestartingSendStreamPreservesRtpState) {
TestRtpStatePreservation(false, false);
}
TEST_P(EndToEndTest, RestartingSendStreamPreservesRtpStatesWithRtx) {
TestRtpStatePreservation(true, false);
}
TEST_P(EndToEndTest, RestartingSendStreamKeepsRtpAndRtcpTimestampsSynced) {
TestRtpStatePreservation(true, true);
}
// This test is flaky on linux_memcheck. Disable on all linux bots until
// flakyness has been fixed.
// https://bugs.chromium.org/p/webrtc/issues/detail?id=7737
#if defined(WEBRTC_LINUX)
TEST_P(EndToEndTest, DISABLED_TestFlexfecRtpStatePreservation) {
#else
TEST_P(EndToEndTest, TestFlexfecRtpStatePreservation) {
#endif
class RtpSequenceObserver : public test::RtpRtcpObserver {
public:
RtpSequenceObserver()
: test::RtpRtcpObserver(kDefaultTimeoutMs),
num_flexfec_packets_sent_(0) {}
void ResetPacketCount() {
rtc::CritScope lock(&crit_);
num_flexfec_packets_sent_ = 0;
}
private:
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
const uint16_t sequence_number = header.sequenceNumber;
const uint32_t timestamp = header.timestamp;
const uint32_t ssrc = header.ssrc;
if (ssrc == kVideoSendSsrcs[0] || ssrc == kSendRtxSsrcs[0]) {
return SEND_PACKET;
}
EXPECT_EQ(kFlexfecSendSsrc, ssrc) << "Unknown SSRC sent.";
++num_flexfec_packets_sent_;
// If this is the first packet, we have nothing to compare to.
if (!last_observed_sequence_number_) {
last_observed_sequence_number_.emplace(sequence_number);
last_observed_timestamp_.emplace(timestamp);
return SEND_PACKET;
}
// Verify continuity and monotonicity of RTP sequence numbers.
EXPECT_EQ(static_cast<uint16_t>(*last_observed_sequence_number_ + 1),
sequence_number);
last_observed_sequence_number_.emplace(sequence_number);
// Timestamps should be non-decreasing...
const bool timestamp_is_same_or_newer =
timestamp == *last_observed_timestamp_ ||
IsNewerTimestamp(timestamp, *last_observed_timestamp_);
EXPECT_TRUE(timestamp_is_same_or_newer);
// ...but reasonably close in time.
const int k10SecondsInRtpTimestampBase = 10 * kVideoPayloadTypeFrequency;
EXPECT_TRUE(IsNewerTimestamp(
*last_observed_timestamp_ + k10SecondsInRtpTimestampBase, timestamp));
last_observed_timestamp_.emplace(timestamp);
// Pass test when enough packets have been let through.
if (num_flexfec_packets_sent_ >= 10) {
observation_complete_.Set();
}
return SEND_PACKET;
}
rtc::Optional<uint16_t> last_observed_sequence_number_
RTC_GUARDED_BY(crit_);
rtc::Optional<uint32_t> last_observed_timestamp_ RTC_GUARDED_BY(crit_);
size_t num_flexfec_packets_sent_ RTC_GUARDED_BY(crit_);
rtc::CriticalSection crit_;
} observer;
static constexpr int kFrameMaxWidth = 320;
static constexpr int kFrameMaxHeight = 180;
static constexpr int kFrameRate = 15;
Call::Config config(event_log_.get());
std::unique_ptr<test::PacketTransport> send_transport;
std::unique_ptr<test::PacketTransport> receive_transport;
std::unique_ptr<VideoEncoder> encoder;
task_queue_.SendTask([&]() {
CreateCalls(config, config);
FakeNetworkPipe::Config lossy_delayed_link;
lossy_delayed_link.loss_percent = 2;
lossy_delayed_link.queue_delay_ms = 50;
send_transport = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, sender_call_.get(), &observer,
test::PacketTransport::kSender, payload_type_map_, lossy_delayed_link);
send_transport->SetReceiver(receiver_call_->Receiver());
FakeNetworkPipe::Config flawless_link;
receive_transport = rtc::MakeUnique<test::PacketTransport>(
&task_queue_, nullptr, &observer, test::PacketTransport::kReceiver,
payload_type_map_, flawless_link);
receive_transport->SetReceiver(sender_call_->Receiver());
// For reduced flakyness, we use a real VP8 encoder together with NACK
// and RTX.
const int kNumVideoStreams = 1;
const int kNumFlexfecStreams = 1;
CreateSendConfig(kNumVideoStreams, 0, kNumFlexfecStreams,
send_transport.get());
encoder = VP8Encoder::Create();
video_send_config_.encoder_settings.encoder = encoder.get();
video_send_config_.encoder_settings.payload_name = "VP8";
video_send_config_.encoder_settings.payload_type = kVideoSendPayloadType;
video_send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_send_config_.rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]);
video_send_config_.rtp.rtx.payload_type = kSendRtxPayloadType;
CreateMatchingReceiveConfigs(receive_transport.get());
video_receive_configs_[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
video_receive_configs_[0].rtp.rtx_ssrc = kSendRtxSsrcs[0];
video_receive_configs_[0]
.rtp.rtx_associated_payload_types[kSendRtxPayloadType] =
kVideoSendPayloadType;
// The matching FlexFEC receive config is not created by
// CreateMatchingReceiveConfigs since this is not a test::BaseTest.
// Set up the receive config manually instead.
FlexfecReceiveStream::Config flexfec_receive_config(
receive_transport.get());
flexfec_receive_config.payload_type =
video_send_config_.rtp.flexfec.payload_type;
flexfec_receive_config.remote_ssrc = video_send_config_.rtp.flexfec.ssrc;
flexfec_receive_config.protected_media_ssrcs =
video_send_config_.rtp.flexfec.protected_media_ssrcs;
flexfec_receive_config.local_ssrc = kReceiverLocalVideoSsrc;
flexfec_receive_config.transport_cc = true;
flexfec_receive_config.rtp_header_extensions.emplace_back(
RtpExtension::kTransportSequenceNumberUri,
test::kTransportSequenceNumberExtensionId);
flexfec_receive_configs_.push_back(flexfec_receive_config);
CreateFlexfecStreams();
CreateVideoStreams();
// RTCP might be disabled if the network is "down".
sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
Start();
});
// Initial test.
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
task_queue_.SendTask([this, &observer]() {
// Ensure monotonicity when the VideoSendStream is restarted.
Stop();
observer.ResetPacketCount();
Start();
});
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
task_queue_.SendTask([this, &observer]() {
// Ensure monotonicity when the VideoSendStream is recreated.
frame_generator_capturer_->Stop();
sender_call_->DestroyVideoSendStream(video_send_stream_);
observer.ResetPacketCount();
video_send_stream_ = sender_call_->CreateVideoSendStream(
video_send_config_.Copy(), video_encoder_config_.Copy());
video_send_stream_->Start();
CreateFrameGeneratorCapturer(kFrameRate, kFrameMaxWidth, kFrameMaxHeight);
frame_generator_capturer_->Start();
});
EXPECT_TRUE(observer.Wait()) << "Timed out waiting for packets.";
// Cleanup.
task_queue_.SendTask([this, &send_transport, &receive_transport]() {
Stop();
DestroyStreams();
send_transport.reset();
receive_transport.reset();
DestroyCalls();
});
}
TEST_P(EndToEndTest, RespectsNetworkState) {
// TODO(pbos): Remove accepted downtime packets etc. when signaling network
// down blocks until no more packets will be sent.
// Pacer will send from its packet list and then send required padding before
// checking paused_ again. This should be enough for one round of pacing,
// otherwise increase.
static const int kNumAcceptedDowntimeRtp = 5;
// A single RTCP may be in the pipeline.
static const int kNumAcceptedDowntimeRtcp = 1;
class NetworkStateTest : public test::EndToEndTest, public test::FakeEncoder {
public:
explicit NetworkStateTest(
test::SingleThreadedTaskQueueForTesting* task_queue)
: EndToEndTest(kDefaultTimeoutMs),
FakeEncoder(Clock::GetRealTimeClock()),
task_queue_(task_queue),
encoded_frames_(false, false),
packet_event_(false, false),
sender_call_(nullptr),
receiver_call_(nullptr),
sender_state_(kNetworkUp),
sender_rtp_(0),
sender_padding_(0),
sender_rtcp_(0),
receiver_rtcp_(0),
down_frames_(0) {}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&test_crit_);
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
if (length == header.headerLength + header.paddingLength)
++sender_padding_;
++sender_rtp_;
packet_event_.Set();
return SEND_PACKET;
}
Action OnSendRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&test_crit_);
++sender_rtcp_;
packet_event_.Set();
return SEND_PACKET;
}
Action OnReceiveRtp(const uint8_t* packet, size_t length) override {
ADD_FAILURE() << "Unexpected receiver RTP, should not be sending.";
return SEND_PACKET;
}
Action OnReceiveRtcp(const uint8_t* packet, size_t length) override {
rtc::CritScope lock(&test_crit_);
++receiver_rtcp_;
packet_event_.Set();
return SEND_PACKET;
}
void OnCallsCreated(Call* sender_call, Call* receiver_call) override {
sender_call_ = sender_call;
receiver_call_ = receiver_call;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->encoder_settings.encoder = this;
}
void PerformTest() override {
EXPECT_TRUE(encoded_frames_.Wait(kDefaultTimeoutMs))
<< "No frames received by the encoder.";
task_queue_->SendTask([this]() {
// Wait for packets from both sender/receiver.
WaitForPacketsOrSilence(false, false);
// Sender-side network down for audio; there should be no effect on
// video
sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkDown);
WaitForPacketsOrSilence(false, false);
// Receiver-side network down for audio; no change expected
receiver_call_->SignalChannelNetworkState(MediaType::AUDIO,
kNetworkDown);
WaitForPacketsOrSilence(false, false);
// Sender-side network down.
sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkDown);
{
rtc::CritScope lock(&test_crit_);
// After network goes down we shouldn't be encoding more frames.
sender_state_ = kNetworkDown;
}
// Wait for receiver-packets and no sender packets.
WaitForPacketsOrSilence(true, false);
// Receiver-side network down.
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO,
kNetworkDown);
WaitForPacketsOrSilence(true, true);
// Network up for audio for both sides; video is still not expected to
// start
sender_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
receiver_call_->SignalChannelNetworkState(MediaType::AUDIO, kNetworkUp);
WaitForPacketsOrSilence(true, true);
// Network back up again for both.
{
rtc::CritScope lock(&test_crit_);
// It's OK to encode frames again, as we're about to bring up the
// network.
sender_state_ = kNetworkUp;
}
sender_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
receiver_call_->SignalChannelNetworkState(MediaType::VIDEO, kNetworkUp);
WaitForPacketsOrSilence(false, false);
// TODO(skvlad): add tests to verify that the audio streams are stopped
// when the network goes down for audio once the workaround in
// paced_sender.cc is removed.
});
}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) override {
{
rtc::CritScope lock(&test_crit_);
if (sender_state_ == kNetworkDown) {
++down_frames_;
EXPECT_LE(down_frames_, 1)
<< "Encoding more than one frame while network is down.";
if (down_frames_ > 1)
encoded_frames_.Set();
} else {
encoded_frames_.Set();
}
}
return test::FakeEncoder::Encode(
input_image, codec_specific_info, frame_types);
}
private:
void WaitForPacketsOrSilence(bool sender_down, bool receiver_down) {
int64_t initial_time_ms = clock_->TimeInMilliseconds();
int initial_sender_rtp;
int initial_sender_rtcp;
int initial_receiver_rtcp;
{
rtc::CritScope lock(&test_crit_);
initial_sender_rtp = sender_rtp_;
initial_sender_rtcp = sender_rtcp_;
initial_receiver_rtcp = receiver_rtcp_;
}
bool sender_done = false;
bool receiver_done = false;
while (!sender_done || !receiver_done) {
packet_event_.Wait(kSilenceTimeoutMs);
int64_t time_now_ms = clock_->TimeInMilliseconds();
rtc::CritScope lock(&test_crit_);
if (sender_down) {
ASSERT_LE(sender_rtp_ - initial_sender_rtp - sender_padding_,
kNumAcceptedDowntimeRtp)
<< "RTP sent during sender-side downtime.";
ASSERT_LE(sender_rtcp_ - initial_sender_rtcp,
kNumAcceptedDowntimeRtcp)
<< "RTCP sent during sender-side downtime.";
if (time_now_ms - initial_time_ms >=
static_cast<int64_t>(kSilenceTimeoutMs)) {
sender_done = true;
}
} else {
if (sender_rtp_ > initial_sender_rtp + kNumAcceptedDowntimeRtp)
sender_done = true;
}
if (receiver_down) {
ASSERT_LE(receiver_rtcp_ - initial_receiver_rtcp,
kNumAcceptedDowntimeRtcp)
<< "RTCP sent during receiver-side downtime.";
if (time_now_ms - initial_time_ms >=
static_cast<int64_t>(kSilenceTimeoutMs)) {
receiver_done = true;
}
} else {
if (receiver_rtcp_ > initial_receiver_rtcp + kNumAcceptedDowntimeRtcp)
receiver_done = true;
}
}
}
test::SingleThreadedTaskQueueForTesting* const task_queue_;
rtc::CriticalSection test_crit_;
rtc::Event encoded_frames_;
rtc::Event packet_event_;
Call* sender_call_;
Call* receiver_call_;
NetworkState sender_state_ RTC_GUARDED_BY(test_crit_);
int sender_rtp_ RTC_GUARDED_BY(test_crit_);
int sender_padding_ RTC_GUARDED_BY(test_crit_);
int sender_rtcp_ RTC_GUARDED_BY(test_crit_);
int receiver_rtcp_ RTC_GUARDED_BY(test_crit_);
int down_frames_ RTC_GUARDED_BY(test_crit_);
} test(&task_queue_);
RunBaseTest(&test);
}
TEST_P(EndToEndTest, CallReportsRttForSender) {
static const int kSendDelayMs = 30;
static const int kReceiveDelayMs = 70;
std::unique_ptr<test::DirectTransport> sender_transport;
std::unique_ptr<test::DirectTransport> receiver_transport;
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
FakeNetworkPipe::Config config;
config.queue_delay_ms = kSendDelayMs;
CreateCalls(Call::Config(event_log_.get()), Call::Config(event_log_.get()));
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, config, sender_call_.get(), payload_type_map_);
config.queue_delay_ms = kReceiveDelayMs;
receiver_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, config, receiver_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
receiver_transport->SetReceiver(sender_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(receiver_transport.get());
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
int64_t start_time_ms = clock_->TimeInMilliseconds();
while (true) {
Call::Stats stats = sender_call_->GetStats();
ASSERT_GE(start_time_ms + kDefaultTimeoutMs,
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);
}
task_queue_.SendTask([this, &sender_transport, &receiver_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
receiver_transport.reset();
DestroyCalls();
});
}
void EndToEndTest::VerifyNewVideoSendStreamsRespectNetworkState(
MediaType network_to_bring_up,
VideoEncoder* encoder,
Transport* transport) {
task_queue_.SendTask([this, network_to_bring_up, encoder, transport]() {
CreateSenderCall(Call::Config(event_log_.get()));
sender_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
CreateSendConfig(1, 0, 0, transport);
video_send_config_.encoder_settings.encoder = encoder;
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
SleepMs(kSilenceTimeoutMs);
task_queue_.SendTask([this]() {
Stop();
DestroyStreams();
DestroyCalls();
});
}
void EndToEndTest::VerifyNewVideoReceiveStreamsRespectNetworkState(
MediaType network_to_bring_up,
Transport* transport) {
std::unique_ptr<test::DirectTransport> sender_transport;
task_queue_.SendTask([this, &sender_transport, network_to_bring_up,
transport]() {
Call::Config config(event_log_.get());
CreateCalls(config, config);
receiver_call_->SignalChannelNetworkState(network_to_bring_up, kNetworkUp);
sender_transport = rtc::MakeUnique<test::DirectTransport>(
&task_queue_, sender_call_.get(), payload_type_map_);
sender_transport->SetReceiver(receiver_call_->Receiver());
CreateSendConfig(1, 0, 0, sender_transport.get());
CreateMatchingReceiveConfigs(transport);
CreateVideoStreams();
CreateFrameGeneratorCapturer(kDefaultFramerate, kDefaultWidth,
kDefaultHeight);
Start();
});
SleepMs(kSilenceTimeoutMs);
task_queue_.SendTask([this, &sender_transport]() {
Stop();
DestroyStreams();
sender_transport.reset();
DestroyCalls();
});
}
TEST_P(EndToEndTest, NewVideoSendStreamsRespectVideoNetworkDown) {
class UnusedEncoder : public test::FakeEncoder {
public:
UnusedEncoder() : FakeEncoder(Clock::GetRealTimeClock()) {}
int32_t InitEncode(const VideoCodec* config,
int32_t number_of_cores,
size_t max_payload_size) override {
EXPECT_GT(config->startBitrate, 0u);
return 0;
}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) override {
ADD_FAILURE() << "Unexpected frame encode.";
return test::FakeEncoder::Encode(input_image, codec_specific_info,
frame_types);
}
};
UnusedEncoder unused_encoder;
UnusedTransport unused_transport;
VerifyNewVideoSendStreamsRespectNetworkState(
MediaType::AUDIO, &unused_encoder, &unused_transport);
}
TEST_P(EndToEndTest, NewVideoSendStreamsIgnoreAudioNetworkDown) {
class RequiredEncoder : public test::FakeEncoder {
public:
RequiredEncoder()
: FakeEncoder(Clock::GetRealTimeClock()), encoded_frame_(false) {}
~RequiredEncoder() {
if (!encoded_frame_) {
ADD_FAILURE() << "Didn't encode an expected frame";
}
}
int32_t Encode(const VideoFrame& input_image,
const CodecSpecificInfo* codec_specific_info,
const std::vector<FrameType>* frame_types) override {
encoded_frame_ = true;
return test::FakeEncoder::Encode(input_image, codec_specific_info,
frame_types);
}
private:
bool encoded_frame_;
};
RequiredTransport required_transport(true /*rtp*/, false /*rtcp*/);
RequiredEncoder required_encoder;
VerifyNewVideoSendStreamsRespectNetworkState(
MediaType::VIDEO, &required_encoder, &required_transport);
}
TEST_P(EndToEndTest, NewVideoReceiveStreamsRespectVideoNetworkDown) {
UnusedTransport transport;
VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::AUDIO, &transport);
}
TEST_P(EndToEndTest, NewVideoReceiveStreamsIgnoreAudioNetworkDown) {
RequiredTransport transport(false /*rtp*/, true /*rtcp*/);
VerifyNewVideoReceiveStreamsRespectNetworkState(MediaType::VIDEO, &transport);
}
void VerifyEmptyNackConfig(const NackConfig& config) {
EXPECT_EQ(0, config.rtp_history_ms)
<< "Enabling NACK requires rtcp-fb: nack negotiation.";
}
void VerifyEmptyUlpfecConfig(const UlpfecConfig& config) {
EXPECT_EQ(-1, config.ulpfec_payload_type)
<< "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
EXPECT_EQ(-1, config.red_payload_type)
<< "Enabling ULPFEC requires rtpmap: red negotiation.";
EXPECT_EQ(-1, config.red_rtx_payload_type)
<< "Enabling RTX in ULPFEC requires rtpmap: rtx negotiation.";
}
void VerifyEmptyFlexfecConfig(
const VideoSendStream::Config::Rtp::Flexfec& config) {
EXPECT_EQ(-1, config.payload_type)
<< "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
EXPECT_EQ(0U, config.ssrc)
<< "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
EXPECT_TRUE(config.protected_media_ssrcs.empty())
<< "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
}
TEST_P(EndToEndTest, VerifyDefaultSendConfigParameters) {
VideoSendStream::Config default_send_config(nullptr);
EXPECT_EQ(0, default_send_config.rtp.nack.rtp_history_ms)
<< "Enabling NACK require rtcp-fb: nack negotiation.";
EXPECT_TRUE(default_send_config.rtp.rtx.ssrcs.empty())
<< "Enabling RTX requires rtpmap: rtx negotiation.";
EXPECT_TRUE(default_send_config.rtp.extensions.empty())
<< "Enabling RTP extensions require negotiation.";
VerifyEmptyNackConfig(default_send_config.rtp.nack);
VerifyEmptyUlpfecConfig(default_send_config.rtp.ulpfec);
VerifyEmptyFlexfecConfig(default_send_config.rtp.flexfec);
}
TEST_P(EndToEndTest, VerifyDefaultVideoReceiveConfigParameters) {
VideoReceiveStream::Config default_receive_config(nullptr);
EXPECT_EQ(RtcpMode::kCompound, default_receive_config.rtp.rtcp_mode)
<< "Reduced-size RTCP require rtcp-rsize to be negotiated.";
EXPECT_FALSE(default_receive_config.rtp.remb)
<< "REMB require rtcp-fb: goog-remb to be negotiated.";
EXPECT_FALSE(
default_receive_config.rtp.rtcp_xr.receiver_reference_time_report)
<< "RTCP XR settings require rtcp-xr to be negotiated.";
EXPECT_EQ(0U, default_receive_config.rtp.rtx_ssrc)
<< "Enabling RTX requires ssrc-group: FID negotiation";
EXPECT_TRUE(default_receive_config.rtp.rtx_associated_payload_types.empty())
<< "Enabling RTX requires rtpmap: rtx negotiation.";
EXPECT_TRUE(default_receive_config.rtp.extensions.empty())
<< "Enabling RTP extensions require negotiation.";
VerifyEmptyNackConfig(default_receive_config.rtp.nack);
EXPECT_EQ(-1, default_receive_config.rtp.ulpfec_payload_type)
<< "Enabling ULPFEC requires rtpmap: ulpfec negotiation.";
EXPECT_EQ(-1, default_receive_config.rtp.red_payload_type)
<< "Enabling ULPFEC requires rtpmap: red negotiation.";
}
TEST_P(EndToEndTest, VerifyDefaultFlexfecReceiveConfigParameters) {
test::NullTransport rtcp_send_transport;
FlexfecReceiveStream::Config default_receive_config(&rtcp_send_transport);
EXPECT_EQ(-1, default_receive_config.payload_type)
<< "Enabling FlexFEC requires rtpmap: flexfec negotiation.";
EXPECT_EQ(0U, default_receive_config.remote_ssrc)
<< "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
EXPECT_TRUE(default_receive_config.protected_media_ssrcs.empty())
<< "Enabling FlexFEC requires ssrc-group: FEC-FR negotiation.";
}
TEST_P(EndToEndTest, TransportSeqNumOnAudioAndVideo) {
static constexpr int kExtensionId = 8;
static constexpr size_t kMinPacketsToWaitFor = 50;
class TransportSequenceNumberTest : public test::EndToEndTest {
public:
TransportSequenceNumberTest()
: EndToEndTest(kDefaultTimeoutMs),
video_observed_(false),
audio_observed_(false) {
parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber,
kExtensionId);
}
size_t GetNumVideoStreams() const override { return 1; }
size_t GetNumAudioStreams() const override { return 1; }
void ModifyAudioConfigs(
AudioSendStream::Config* send_config,
std::vector<AudioReceiveStream::Config>* receive_configs) override {
send_config->rtp.extensions.clear();
send_config->rtp.extensions.push_back(RtpExtension(
RtpExtension::kTransportSequenceNumberUri, kExtensionId));
(*receive_configs)[0].rtp.extensions.clear();
(*receive_configs)[0].rtp.extensions = send_config->rtp.extensions;
}
Action OnSendRtp(const uint8_t* packet, size_t length) override {
RTPHeader header;
EXPECT_TRUE(parser_->Parse(packet, length, &header));
EXPECT_TRUE(header.extension.hasTransportSequenceNumber);
// Unwrap packet id and verify uniqueness.
int64_t packet_id =
unwrapper_.Unwrap(header.extension.transportSequenceNumber);
EXPECT_TRUE(received_packet_ids_.insert(packet_id).second);
if (header.ssrc == kVideoSendSsrcs[0])
video_observed_ = true;
if (header.ssrc == kAudioSendSsrc)
audio_observed_ = true;
if (audio_observed_ && video_observed_ &&
received_packet_ids_.size() >= kMinPacketsToWaitFor) {
size_t packet_id_range =
*received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1;
EXPECT_EQ(received_packet_ids_.size(), packet_id_range);
observation_complete_.Set();
}
return SEND_PACKET;
}
void PerformTest() override {
EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video "
"packets with transport sequence number.";
}
void ExpectSuccessful() {
EXPECT_TRUE(video_observed_);
EXPECT_TRUE(audio_observed_);
EXPECT_GE(received_packet_ids_.size(), kMinPacketsToWaitFor);
}
private:
bool video_observed_;
bool audio_observed_;
SequenceNumberUnwrapper unwrapper_;
std::set<int64_t> received_packet_ids_;
} test;
RunBaseTest(&test);
// Double check conditions for successful test to produce better error
// message when the test fail.
test.ExpectSuccessful();
}
class EndToEndLogTest : public EndToEndTest {
void SetUp() { paths_.clear(); }
void TearDown() {
for (const auto& path : paths_) {
rtc::RemoveFile(path);
}
}
public:
int AddFile() {
paths_.push_back(test::TempFilename(test::OutputPath(), "test_file"));
return static_cast<int>(paths_.size()) - 1;
}
rtc::PlatformFile OpenFile(int idx) {
return rtc::OpenPlatformFile(paths_[idx]);
}
void LogSend(bool open) {
if (open) {
video_send_stream_->EnableEncodedFrameRecording(
std::vector<rtc::PlatformFile>(1, OpenFile(AddFile())), 0);
} else {
video_send_stream_->DisableEncodedFrameRecording();
}
}
void LogReceive(bool open) {
if (open) {
video_receive_streams_[0]->EnableEncodedFrameRecording(
OpenFile(AddFile()), 0);
} else {
video_receive_streams_[0]->DisableEncodedFrameRecording();
}
}
std::vector<std::string> paths_;
};
TEST_P(EndToEndLogTest, LogsEncodedFramesWhenRequested) {
static const int kNumFramesToRecord = 10;
class LogEncodingObserver : public test::EndToEndTest,
public EncodedFrameObserver {
public:
explicit LogEncodingObserver(EndToEndLogTest* fixture)
: EndToEndTest(kDefaultTimeoutMs),
fixture_(fixture),
recorded_frames_(0) {}
void PerformTest() override {
fixture_->LogSend(true);
fixture_->LogReceive(true);
ASSERT_TRUE(Wait()) << "Timed out while waiting for frame logging.";
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStream::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
encoder_ = VP8Encoder::Create();
decoder_ = VP8Decoder::Create();
send_config->post_encode_callback = this;
send_config->encoder_settings.payload_name = "VP8";
send_config->encoder_settings.encoder = encoder_.get();
(*receive_configs)[0].decoders.resize(1);
(*receive_configs)[0].decoders[0].payload_type =
send_config->encoder_settings.payload_type;
(*receive_configs)[0].decoders[0].payload_name =
send_config->encoder_settings.payload_name;
(*receive_configs)[0].decoders[0].decoder = decoder_.get();
}
void EncodedFrameCallback(const EncodedFrame& encoded_frame) override {
rtc::CritScope lock(&crit_);
if (recorded_frames_++ > kNumFramesToRecord) {
fixture_->LogSend(false);
fixture_->LogReceive(false);
rtc::File send_file(fixture_->OpenFile(0));
rtc::File receive_file(fixture_->OpenFile(1));
uint8_t out[100];
// If logging has worked correctly neither file should be empty, i.e.
// we should be able to read something from them.
EXPECT_LT(0u, send_file.Read(out, 100));
EXPECT_LT(0u, receive_file.Read(out, 100));
observation_complete_.Set();
}
}
private:
EndToEndLogTest* const fixture_;
std::unique_ptr<VideoEncoder> encoder_;
std::unique_ptr<VideoDecoder> decoder_;
rtc::CriticalSection crit_;
int recorded_frames_ RTC_GUARDED_BY(crit_);
} test(this);
RunBaseTest(&test);
}
INSTANTIATE_TEST_CASE_P(RoundRobin,
EndToEndTest,
::testing::Values("WebRTC-RoundRobinPacing/Disabled/",
"WebRTC-RoundRobinPacing/Enabled/"));
} // namespace webrtc