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webrtc / src / webrtc / 6ae826202357025d9cb538436a309b7c84449cb2 / . / video / end_to_end_tests.cc
blob: 97b9ee7e35ffed98adfba7d8448f17d738359d5c [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 "webrtc/api/optional.h"
#include "webrtc/api/video_codecs/video_encoder.h"
#include "webrtc/call/call.h"
#include "webrtc/common_video/include/frame_callback.h"
#include "webrtc/logging/rtc_event_log/rtc_event_log.h"
#include "webrtc/media/base/fakevideorenderer.h"
#include "webrtc/media/base/mediaconstants.h"
#include "webrtc/media/engine/internalencoderfactory.h"
#include "webrtc/media/engine/simulcast_encoder_adapter.h"
#include "webrtc/media/engine/webrtcvideoencoderfactory.h"
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_format.h"
#include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
#include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
#include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
#include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
#include "webrtc/modules/video_coding/include/video_coding_defines.h"
#include "webrtc/rtc_base/checks.h"
#include "webrtc/rtc_base/event.h"
#include "webrtc/rtc_base/file.h"
#include "webrtc/rtc_base/ptr_util.h"
#include "webrtc/rtc_base/random.h"
#include "webrtc/rtc_base/rate_limiter.h"
#include "webrtc/system_wrappers/include/metrics.h"
#include "webrtc/system_wrappers/include/metrics_default.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/call_test.h"
#include "webrtc/test/direct_transport.h"
#include "webrtc/test/encoder_settings.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
#include "webrtc/test/field_trial.h"
#include "webrtc/test/frame_generator.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/gmock.h"
#include "webrtc/test/gtest.h"
#include "webrtc/test/null_transport.h"
#include "webrtc/test/rtcp_packet_parser.h"
#include "webrtc/test/rtp_rtcp_observer.h"
#include "webrtc/test/testsupport/fileutils.h"
#include "webrtc/test/testsupport/perf_test.h"
#include "webrtc/video/transport_adapter.h"
// Flaky under MemorySanitizer: bugs.webrtc.org/7419
#if defined(MEMORY_SANITIZER)
#define MAYBE_InitialProbing DISABLED_InitialProbing
// Fails on iOS bots: bugs.webrtc.org/7851
#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
#define MAYBE_InitialProbing DISABLED_InitialProbing
#else
#define MAYBE_InitialProbing InitialProbing
#endif
namespace webrtc {
namespace {
constexpr int kSilenceTimeoutMs = 2000;
}
class EndToEndTest : public test::CallTest {
public:
EndToEndTest() {}
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_red, bool screenshare);
void VerifyNewVideoSendStreamsRespectNetworkState(
MediaType network_to_bring_up,
VideoEncoder* encoder,
Transport* transport);
void VerifyNewVideoReceiveStreamsRespectNetworkState(
MediaType network_to_bring_up,
Transport* transport);
};
TEST_F(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_F(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_F(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_F(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_F(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,
webrtc::VideoEncoder* encoder,
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_(encoder),
decoder_(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_F(EndToEndTest, SendsAndReceivesVP8) {
CodecObserver test(5, kVideoRotation_0, "VP8", VP8Encoder::Create(),
VP8Decoder::Create());
RunBaseTest(&test);
}
TEST_F(EndToEndTest, SendsAndReceivesVP8Rotation90) {
CodecObserver test(5, kVideoRotation_90, "VP8", VP8Encoder::Create(),
VP8Decoder::Create());
RunBaseTest(&test);
}
#if !defined(RTC_DISABLE_VP9)
TEST_F(EndToEndTest, SendsAndReceivesVP9) {
CodecObserver test(500, kVideoRotation_0, "VP9", VP9Encoder::Create(),
VP9Decoder::Create());
RunBaseTest(&test);
}
TEST_F(EndToEndTest, SendsAndReceivesVP9VideoRotation90) {
CodecObserver test(5, kVideoRotation_90, "VP9", VP9Encoder::Create(),
VP9Decoder::Create());
RunBaseTest(&test);
}
#endif // !defined(RTC_DISABLE_VP9)
#if defined(WEBRTC_USE_H264)
TEST_F(EndToEndTest, SendsAndReceivesH264) {
CodecObserver test(500, kVideoRotation_0, "H264",
H264Encoder::Create(cricket::VideoCodec("H264")),
H264Decoder::Create());
RunBaseTest(&test);
}
TEST_F(EndToEndTest, SendsAndReceivesH264VideoRotation90) {
CodecObserver test(5, kVideoRotation_90, "H264",
H264Encoder::Create(cricket::VideoCodec("H264")),
H264Decoder::Create());
RunBaseTest(&test);
}
TEST_F(EndToEndTest, 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_F(EndToEndTest, 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_F(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_F(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_ GUARDED_BY(&crit_);
} test;
RunBaseTest(&test);
}
TEST_F(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_ =
rtc::Optional<uint16_t>(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_F(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.ulpfec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.ulpfec.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_ GUARDED_BY(crit_);
// Several packets can have the same timestamp.
std::multiset<uint32_t> dropped_timestamps_ GUARDED_BY(crit_);
Random random_;
int num_packets_sent_ 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.ulpfec.red_rtx_payload_type =
test::CallTest::kRtxRedPayloadType;
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.ulpfec.red_rtx_payload_type =
test::CallTest::kRtxRedPayloadType;
(*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_ GUARDED_BY(crit_);
// Several packets can have the same timestamp.
std::multiset<uint32_t> dropped_timestamps_ GUARDED_BY(crit_);
const bool enable_nack_;
const bool expect_flexfec_rtcp_;
bool received_flexfec_rtcp_ GUARDED_BY(crit_);
Random random_;
int num_packets_sent_;
};
TEST_F(EndToEndTest, RecoversWithFlexfec) {
FlexfecRenderObserver test(false, false);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, RecoversWithFlexfecAndNack) {
FlexfecRenderObserver test(true, false);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, RecoversWithFlexfecAndSendsCorrespondingRtcp) {
FlexfecRenderObserver test(false, true);
RunBaseTest(&test);
}
TEST_F(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.ulpfec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.ulpfec.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_ 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.ulpfec_payload_type =
send_config->rtp.ulpfec.ulpfec_payload_type;
(*receive_configs)[0].rtp.ulpfec.red_payload_type =
send_config->rtp.ulpfec.red_payload_type;
(*receive_configs)[0].rtp.ulpfec.red_rtx_payload_type =
send_config->rtp.ulpfec.red_rtx_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_red) {
if (use_red) {
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_ GUARDED_BY(&crit_);
std::vector<uint32_t> rendered_timestamps_ GUARDED_BY(&crit_);
} test(enable_rtx, enable_red);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrame) {
DecodesRetransmittedFrame(false, false);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameOverRtx) {
DecodesRetransmittedFrame(true, false);
}
TEST_F(EndToEndTest, DecodesRetransmittedFrameByRed) {
DecodesRetransmittedFrame(false, true);
}
TEST_F(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_ GUARDED_BY(&crit_);
int frames_to_drop_ GUARDED_BY(&crit_);
bool received_pli_ GUARDED_BY(&crit_);
} test(rtp_history_ms);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, ReceivesPliAndRecoversWithNack) {
ReceivesPliAndRecovers(1000);
}
TEST_F(EndToEndTest, ReceivesPliAndRecoversWithoutNack) {
ReceivesPliAndRecovers(0);
}
TEST_F(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,
const uint8_t* packet,
size_t length,
const PacketTime& packet_time) override {
if (RtpHeaderParser::IsRtcp(packet, length)) {
return receiver_->DeliverPacket(media_type, packet, length,
packet_time);
} else {
DeliveryStatus delivery_status =
receiver_->DeliverPacket(media_type, packet, length, 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_ GUARDED_BY(&crit_);
int sent_rtcp_ GUARDED_BY(&crit_);
} test(rtcp_mode);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, UsesRtcpCompoundMode) {
RespectsRtcpMode(RtcpMode::kCompound);
}
TEST_F(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].reset(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_F(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_F(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_F(EndToEndTest, VideoReceivesTransportFeedback) {
TransportFeedbackTester test(true, 1, 0);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, VideoTransportFeedbackNotConfigured) {
TransportFeedbackTester test(false, 1, 0);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, AudioReceivesTransportFeedback) {
TransportFeedbackTester test(true, 0, 1);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, AudioTransportFeedbackNotConfigured) {
TransportFeedbackTester test(false, 0, 1);
RunBaseTest(&test);
}
TEST_F(EndToEndTest, AudioVideoReceivesTransportFeedback) {
TransportFeedbackTester test(true, 1, 1);
RunBaseTest(&test);
}
TEST_F(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_ GUARDED_BY(crit_);
int padding_sent_ GUARDED_BY(crit_);
} test(1, 0);
RunBaseTest(&test);
}
TEST_F(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_F(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_F(EndToEndTest, VerifySendSideBweStats) {
BandwidthStatsTest test(true);
RunBaseTest(&test);
}
TEST_F(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_F(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_->SetREMBStatus(true);
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_->SetREMBData(
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_->SetREMBData(
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_F(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_;
};
TEST_F(EndToEndTest, MAYBE_InitialProbing) {
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_F(EndToEndTest, DISABLED_TriggerMidCallProbing) {
// Fails on iOS bots: bugs.webrtc.org/7851
#elif defined(TARGET_IPHONE_SIMULATOR) && TARGET_IPHONE_SIMULATOR
TEST_F(EndToEndTest, DISABLED_TriggerMidCallProbing) {
#else
TEST_F(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_F(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() 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_ GUARDED_BY(&crit_);
bool dropped_rtp_packet_requested_ 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_red,
bool screenshare) {
class StatsObserver : public test::EndToEndTest,
public rtc::VideoSinkInterface<VideoFrame> {
public:
StatsObserver(bool use_rtx, bool use_red, bool screenshare)
: EndToEndTest(kLongTimeoutMs),
use_rtx_(use_rtx),
use_red_(use_red),
screenshare_(screenshare),
// This test uses NACK, so to send FEC we can't use a fake encoder.
vp8_encoder_(use_red ? 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_red_) {
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.ulpfec.red_payload_type = kRedPayloadType;
(*receive_configs)[0].rtp.ulpfec.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;
}
// 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_red_;
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_ GUARDED_BY(&crit_);
} test(use_rtx, use_red, 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"));