diff options
Diffstat (limited to 'third_party/libwebrtc/video/end_to_end_tests/retransmission_tests.cc')
| -rw-r--r-- | third_party/libwebrtc/video/end_to_end_tests/retransmission_tests.cc | 513 |
1 files changed, 513 insertions, 0 deletions
diff --git a/third_party/libwebrtc/video/end_to_end_tests/retransmission_tests.cc b/third_party/libwebrtc/video/end_to_end_tests/retransmission_tests.cc new file mode 100644 index 0000000000..45a9dae1e8 --- /dev/null +++ b/third_party/libwebrtc/video/end_to_end_tests/retransmission_tests.cc @@ -0,0 +1,513 @@ +/* + * 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 "call/simulated_network.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/rtcp_packet_parser.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(kLongTimeout), + sent_rtp_packets_(0), + packets_left_to_drop_(0), + nacks_left_(kNumberOfNacksToObserve) {} + + private: + Action OnSendRtp(const uint8_t* packet, size_t length) override { + MutexLock lock(&mutex_); + RtpPacket rtp_packet; + EXPECT_TRUE(rtp_packet.Parse(packet, length)); + + // 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(const uint8_t* packet, size_t length) override { + MutexLock lock(&mutex_); + 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<VideoReceiveStreamInterface::Config>* receive_configs, + VideoEncoderConfig* encoder_config) override { + send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs; + (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs; + } + + void 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(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(const uint8_t* packet, size_t length) override { + RtpPacket rtp_packet; + EXPECT_TRUE(rtp_packet.Parse(packet, length)); + + 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.data(), buffer.size())); + } + + return SEND_PACKET; + } + + void ModifyAudioConfigs(AudioSendStream::Config* send_config, + std::vector<AudioReceiveStreamInterface::Config>* + receive_configs) override { + (*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; + 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_; + absl::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(const uint8_t* packet, size_t length) override { + test::RtcpPacketParser parser; + EXPECT_TRUE(parser.Parse(packet, length)); + 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(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(const uint8_t* packet, size_t length) override { + MutexLock lock(&mutex_); + RtpPacket rtp_packet; + EXPECT_TRUE(rtp_packet.Parse(packet, length)); + + // 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(const uint8_t* packet, size_t length) override { + MutexLock lock(&mutex_); + 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 { + MutexLock lock(&mutex_); + 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<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(kDefaultTimeout), + payload_type_(GetPayloadType(false, enable_red)), + retransmission_ssrc_(enable_rtx ? kSendRtxSsrcs[0] + : kVideoSendSsrcs[0]), + retransmission_payload_type_(GetPayloadType(enable_rtx, enable_red)), + encoder_factory_([]() { return VP8Encoder::Create(); }), + marker_bits_observed_(0), + retransmitted_timestamp_(0) {} + + private: + Action OnSendRtp(const uint8_t* packet, size_t length) override { + MutexLock lock(&mutex_); + RtpPacket rtp_packet; + EXPECT_TRUE(rtp_packet.Parse(packet, length)); + + // 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.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 = 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 = kNackRtpHistoryMs; + + if (payload_type_ == kRedPayloadType) { + send_config->rtp.ulpfec.ulpfec_payload_type = kUlpfecPayloadType; + send_config->rtp.ulpfec.red_payload_type = kRedPayloadType; + if (retransmission_ssrc_ == kSendRtxSsrcs[0]) + send_config->rtp.ulpfec.red_rtx_payload_type = kRtxRedPayloadType; + (*receive_configs)[0].rtp.ulpfec_payload_type = + send_config->rtp.ulpfec.ulpfec_payload_type; + (*receive_configs)[0].rtp.red_payload_type = + send_config->rtp.ulpfec.red_payload_type; + } + + if (retransmission_ssrc_ == kSendRtxSsrcs[0]) { + send_config->rtp.rtx.ssrcs.push_back(kSendRtxSsrcs[0]); + send_config->rtp.rtx.payload_type = kSendRtxPayloadType; + (*receive_configs)[0].rtp.rtx_ssrc = kSendRtxSsrcs[0]; + (*receive_configs)[0] + .rtp.rtx_associated_payload_types[(payload_type_ == kRedPayloadType) + ? kRtxRedPayloadType + : kSendRtxPayloadType] = + payload_type_; + } + // Configure encoding and decoding with VP8, since generic packetization + // doesn't support FEC with NACK. + RTC_DCHECK_EQ(1, (*receive_configs)[0].decoders.size()); + send_config->encoder_settings.encoder_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 kRtxRedPayloadType; + return kRedPayloadType; + } + if (use_rtx) + return kSendRtxPayloadType; + return 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 |