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/*
* Copyright 2018 The WebRTC project authors. All Rights Reserved.
*
* Use of this source code is governed by a BSD-style license
* that can be found in the LICENSE file in the root of the source
* tree. An additional intellectual property rights grant can be found
* in the file PATENTS. All contributing project authors may
* be found in the AUTHORS file in the root of the source tree.
*/
#include <memory>
#include "absl/algorithm/container.h"
#include "api/task_queue/task_queue_base.h"
#include "api/test/simulated_network.h"
#include "api/test/video/function_video_encoder_factory.h"
#include "api/units/time_delta.h"
#include "call/fake_network_pipe.h"
#include "modules/rtp_rtcp/source/rtp_packet.h"
#include "modules/video_coding/codecs/vp8/include/vp8.h"
#include "rtc_base/event.h"
#include "rtc_base/synchronization/mutex.h"
#include "rtc_base/task_queue_for_test.h"
#include "test/call_test.h"
#include "test/field_trial.h"
#include "test/gtest.h"
#include "test/network/simulated_network.h"
#include "test/rtcp_packet_parser.h"
#include "test/video_test_constants.h"
namespace webrtc {
namespace {
enum : int { // The first valid value is 1.
kVideoRotationExtensionId = 1,
};
} // namespace
class RetransmissionEndToEndTest : public test::CallTest {
public:
RetransmissionEndToEndTest() {
RegisterRtpExtension(RtpExtension(RtpExtension::kVideoRotationUri,
kVideoRotationExtensionId));
}
protected:
void DecodesRetransmittedFrame(bool enable_rtx, bool enable_red);
void ReceivesPliAndRecovers(int rtp_history_ms);
};
TEST_F(RetransmissionEndToEndTest, ReceivesAndRetransmitsNack) {
static const int kNumberOfNacksToObserve = 2;
static const int kLossBurstSize = 2;
static const int kPacketsBetweenLossBursts = 9;
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(test::VideoTestConstants::kLongTimeout),
sent_rtp_packets_(0),
packets_left_to_drop_(0),
nacks_left_(kNumberOfNacksToObserve) {}
private:
Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet));
// Never drop retransmitted packets.
if (dropped_packets_.find(rtp_packet.SequenceNumber()) !=
dropped_packets_.end()) {
retransmitted_packets_.insert(rtp_packet.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 && rtp_packet.padding_size() == 0) {
--packets_left_to_drop_;
dropped_packets_.insert(rtp_packet.SequenceNumber());
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(rtc::ArrayView<const uint8_t> packet) override {
MutexLock lock(&mutex_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet));
nacks_left_ -= parser.nack()->num_packets();
return SEND_PACKET;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms =
test::VideoTestConstants::kNackRtpHistoryMs;
(*receive_configs)[0].rtp.nack.rtp_history_ms =
test::VideoTestConstants::kNackRtpHistoryMs;
}
void PerformTest() override {
EXPECT_TRUE(Wait())
<< "Timed out waiting for packets to be NACKed, retransmitted and "
"rendered.";
}
Mutex mutex_;
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(&mutex_);
} test;
RunBaseTest(&test);
}
TEST_F(RetransmissionEndToEndTest, ReceivesNackAndRetransmitsAudio) {
class NackObserver : public test::EndToEndTest {
public:
NackObserver()
: EndToEndTest(test::VideoTestConstants::kLongTimeout),
local_ssrc_(0),
remote_ssrc_(0),
receive_transport_(nullptr) {}
private:
size_t GetNumVideoStreams() const override { return 0; }
size_t GetNumAudioStreams() const override { return 1; }
Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet));
if (!sequence_number_to_retransmit_) {
sequence_number_to_retransmit_ = rtp_packet.SequenceNumber();
return DROP_PACKET;
// Don't ask for retransmission straight away, may be deduped in pacer.
} else if (rtp_packet.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));
}
return SEND_PACKET;
}
void ModifyAudioConfigs(AudioSendStream::Config* send_config,
std::vector<AudioReceiveStreamInterface::Config>*
receive_configs) override {
(*receive_configs)[0].rtp.nack.rtp_history_ms =
test::VideoTestConstants::kNackRtpHistoryMs;
local_ssrc_ = (*receive_configs)[0].rtp.local_ssrc;
remote_ssrc_ = (*receive_configs)[0].rtp.remote_ssrc;
receive_transport_ = (*receive_configs)[0].rtcp_send_transport;
}
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_;
std::optional<uint16_t> sequence_number_to_retransmit_;
} test;
RunBaseTest(&test);
}
TEST_F(RetransmissionEndToEndTest,
StopSendingKeyframeRequestsForInactiveStream) {
class KeyframeRequestObserver : public test::EndToEndTest {
public:
explicit KeyframeRequestObserver(TaskQueueBase* task_queue)
: clock_(Clock::GetRealTimeClock()), task_queue_(task_queue) {}
void OnVideoStreamsCreated(VideoSendStream* send_stream,
const std::vector<VideoReceiveStreamInterface*>&
receive_streams) override {
RTC_DCHECK_EQ(1, receive_streams.size());
send_stream_ = send_stream;
receive_stream_ = receive_streams[0];
}
Action OnReceiveRtcp(rtc::ArrayView<const uint8_t> packet) override {
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet));
if (parser.pli()->num_packets() > 0)
task_queue_->PostTask([this] { Run(); });
return SEND_PACKET;
}
bool PollStats() {
if (receive_stream_->GetStats().frames_decoded > 0) {
frame_decoded_ = true;
} else if (clock_->TimeInMilliseconds() - start_time_ < 5000) {
task_queue_->PostDelayedTask([this] { Run(); }, TimeDelta::Millis(100));
return false;
}
return true;
}
void PerformTest() override {
start_time_ = clock_->TimeInMilliseconds();
task_queue_->PostTask([this] { Run(); });
test_done_.Wait(rtc::Event::kForever);
}
void Run() {
if (!frame_decoded_) {
if (PollStats()) {
send_stream_->Stop();
if (!frame_decoded_) {
test_done_.Set();
} else {
// Now we wait for the PLI packet. Once we receive it, a task
// will be posted (see OnReceiveRtcp) and we'll check the stats
// once more before signaling that we're done.
}
}
} else {
EXPECT_EQ(
1U,
receive_stream_->GetStats().rtcp_packet_type_counts.pli_packets);
test_done_.Set();
}
}
private:
Clock* const clock_;
VideoSendStream* send_stream_;
VideoReceiveStreamInterface* receive_stream_;
TaskQueueBase* const task_queue_;
rtc::Event test_done_;
bool frame_decoded_ = false;
int64_t start_time_ = 0;
} test(task_queue());
RunBaseTest(&test);
}
void RetransmissionEndToEndTest::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(test::VideoTestConstants::kLongTimeout),
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(rtc::ArrayView<const uint8_t> packet) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet));
// Drop all retransmitted packets to force a PLI.
if (rtp_packet.Timestamp() <= highest_dropped_timestamp_)
return DROP_PACKET;
if (frames_to_drop_ > 0) {
highest_dropped_timestamp_ = rtp_packet.Timestamp();
--frames_to_drop_;
return DROP_PACKET;
}
return SEND_PACKET;
}
Action OnReceiveRtcp(rtc::ArrayView<const uint8_t> packet) override {
MutexLock lock(&mutex_);
test::RtcpPacketParser parser;
EXPECT_TRUE(parser.Parse(packet));
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 {
MutexLock lock(&mutex_);
if (received_pli_ &&
video_frame.rtp_timestamp() > highest_dropped_timestamp_) {
observation_complete_.Set();
}
if (!received_pli_)
frames_to_drop_ = kPacketsToDrop;
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::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.";
}
Mutex mutex_;
int rtp_history_ms_;
bool nack_enabled_;
uint32_t highest_dropped_timestamp_ RTC_GUARDED_BY(&mutex_);
int frames_to_drop_ RTC_GUARDED_BY(&mutex_);
bool received_pli_ RTC_GUARDED_BY(&mutex_);
} test(rtp_history_ms);
RunBaseTest(&test);
}
TEST_F(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithNack) {
ReceivesPliAndRecovers(1000);
}
TEST_F(RetransmissionEndToEndTest, ReceivesPliAndRecoversWithoutNack) {
ReceivesPliAndRecovers(0);
}
// This test drops second RTP packet with a marker bit set, makes sure it's
// retransmitted and renders. Retransmission SSRCs are also checked.
void RetransmissionEndToEndTest::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(test::VideoTestConstants::kDefaultTimeout),
payload_type_(GetPayloadType(false, enable_red)),
retransmission_ssrc_(
enable_rtx ? test::VideoTestConstants::kSendRtxSsrcs[0]
: test::VideoTestConstants::kVideoSendSsrcs[0]),
retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)),
encoder_factory_(
[](const Environment& env, const SdpVideoFormat& format) {
return CreateVp8Encoder(env);
}),
marker_bits_observed_(0),
retransmitted_timestamp_(0) {}
private:
Action OnSendRtp(rtc::ArrayView<const uint8_t> packet) override {
MutexLock lock(&mutex_);
RtpPacket rtp_packet;
EXPECT_TRUE(rtp_packet.Parse(packet));
// Ignore padding-only packets over RTX.
if (rtp_packet.PayloadType() != payload_type_) {
EXPECT_EQ(retransmission_ssrc_, rtp_packet.Ssrc());
if (rtp_packet.payload_size() == 0)
return SEND_PACKET;
}
if (rtp_packet.Timestamp() == retransmitted_timestamp_) {
EXPECT_EQ(retransmission_ssrc_, rtp_packet.Ssrc());
EXPECT_EQ(retransmission_payload_type_, rtp_packet.PayloadType());
return SEND_PACKET;
}
// Found the final packet of the frame to inflict loss to, drop this and
// expect a retransmission.
if (rtp_packet.PayloadType() == payload_type_ && rtp_packet.Marker() &&
++marker_bits_observed_ == kDroppedFrameNumber) {
// This should be the only dropped packet.
EXPECT_EQ(0u, retransmitted_timestamp_);
retransmitted_timestamp_ = rtp_packet.Timestamp();
return DROP_PACKET;
}
return SEND_PACKET;
}
void OnFrame(const VideoFrame& frame) override {
EXPECT_EQ(kVideoRotation_90, frame.rotation());
{
MutexLock lock(&mutex_);
if (frame.rtp_timestamp() == retransmitted_timestamp_)
observation_complete_.Set();
}
orig_renderer_->OnFrame(frame);
}
void ModifyVideoConfigs(
VideoSendStream::Config* send_config,
std::vector<VideoReceiveStreamInterface::Config>* receive_configs,
VideoEncoderConfig* encoder_config) override {
send_config->rtp.nack.rtp_history_ms =
test::VideoTestConstants::kNackRtpHistoryMs;
// Insert ourselves into the rendering pipeline.
RTC_DCHECK(!orig_renderer_);
orig_renderer_ = (*receive_configs)[0].renderer;
RTC_DCHECK(orig_renderer_);
// To avoid post-decode frame dropping, disable the prerender buffer.
(*receive_configs)[0].enable_prerenderer_smoothing = false;
(*receive_configs)[0].renderer = this;
(*receive_configs)[0].rtp.nack.rtp_history_ms =
test::VideoTestConstants::kNackRtpHistoryMs;
if (payload_type_ == test::VideoTestConstants::kRedPayloadType) {
send_config->rtp.ulpfec.ulpfec_payload_type =
test::VideoTestConstants::kUlpfecPayloadType;
send_config->rtp.ulpfec.red_payload_type =
test::VideoTestConstants::kRedPayloadType;
if (retransmission_ssrc_ == test::VideoTestConstants::kSendRtxSsrcs[0])
send_config->rtp.ulpfec.red_rtx_payload_type =
test::VideoTestConstants::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_ == test::VideoTestConstants::kSendRtxSsrcs[0]) {
send_config->rtp.rtx.ssrcs.push_back(
test::VideoTestConstants::kSendRtxSsrcs[0]);
send_config->rtp.rtx.payload_type =
test::VideoTestConstants::kSendRtxPayloadType;
(*receive_configs)[0].rtp.rtx_ssrc =
test::VideoTestConstants::kSendRtxSsrcs[0];
(*receive_configs)[0].rtp.rtx_associated_payload_types
[(payload_type_ == test::VideoTestConstants::kRedPayloadType)
? test::VideoTestConstants::kRtxRedPayloadType
: test::VideoTestConstants::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_factory = &encoder_factory_;
send_config->rtp.payload_name = "VP8";
encoder_config->codec_type = kVideoCodecVP8;
(*receive_configs)[0].decoders[0].video_format = SdpVideoFormat::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 test::VideoTestConstants::kRtxRedPayloadType;
return test::VideoTestConstants::kRedPayloadType;
}
if (use_rtx)
return test::VideoTestConstants::kSendRtxPayloadType;
return test::VideoTestConstants::kFakeVideoSendPayloadType;
}
Mutex mutex_;
rtc::VideoSinkInterface<VideoFrame>* orig_renderer_ = nullptr;
const int payload_type_;
const uint32_t retransmission_ssrc_;
const int retransmission_payload_type_;
test::FunctionVideoEncoderFactory encoder_factory_;
const std::string payload_name_;
int marker_bits_observed_;
uint32_t retransmitted_timestamp_ RTC_GUARDED_BY(&mutex_);
} test(enable_rtx, enable_red);
RunBaseTest(&test);
}
TEST_F(RetransmissionEndToEndTest, DecodesRetransmittedFrame) {
DecodesRetransmittedFrame(false, false);
}
TEST_F(RetransmissionEndToEndTest, DecodesRetransmittedFrameOverRtx) {
DecodesRetransmittedFrame(true, false);
}
TEST_F(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRed) {
DecodesRetransmittedFrame(false, true);
}
TEST_F(RetransmissionEndToEndTest, DecodesRetransmittedFrameByRedOverRtx) {
DecodesRetransmittedFrame(true, true);
}
} // namespace webrtc