| /* |
| * 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 "absl/memory/memory.h" |
| #include "call/call.h" |
| #include "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "test/call_test.h" |
| #include "test/field_trial.h" |
| #include "test/gtest.h" |
| |
| #include "modules/rtp_rtcp/source/byte_io.h" |
| #include "test/rtcp_packet_parser.h" |
| #include "video/end_to_end_tests/multi_stream_tester.h" |
| |
| namespace webrtc { |
| namespace { |
| enum : int { // The first valid value is 1. |
| kTransportSequenceNumberExtensionId = 1, |
| }; |
| } // namespace |
| |
| class TransportFeedbackEndToEndTest : public test::CallTest { |
| public: |
| TransportFeedbackEndToEndTest() { |
| RegisterRtpExtension(RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberExtensionId)); |
| } |
| }; |
| |
| TEST_F(TransportFeedbackEndToEndTest, AssignsTransportSequenceNumbers) { |
| 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, |
| absl::make_unique<FakeNetworkPipe>( |
| Clock::GetRealTimeClock(), |
| absl::make_unique<SimulatedNetwork>( |
| BuiltInNetworkBehaviorConfig())), |
| sender_call, |
| payload_type_map), |
| 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, |
| kTransportSequenceNumberExtensionId); |
| } |
| 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() == MultiStreamTester::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 MultiStreamTester { |
| public: |
| explicit TransportSequenceNumberTester( |
| test::SingleThreadedTaskQueueForTesting* task_queue) |
| : MultiStreamTester(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, |
| kTransportSequenceNumberExtensionId)); |
| |
| // 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, |
| kTransportSequenceNumberExtensionId)); |
| receive_config->renderer = &fake_renderer_; |
| } |
| |
| test::DirectTransport* CreateSendTransport( |
| test::SingleThreadedTaskQueueForTesting* task_queue, |
| Call* sender_call) override { |
| std::map<uint8_t, MediaType> payload_type_map = |
| MultiStreamTester::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::TransportFeedbackEndToEndTest::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, |
| kTransportSequenceNumberExtensionId)); |
| (*receive_configs)[0].rtp.extensions.clear(); |
| (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; |
| (*receive_configs)[0].rtp.transport_cc = feedback_enabled_; |
| } |
| |
| private: |
| const bool feedback_enabled_; |
| const size_t num_video_streams_; |
| const size_t num_audio_streams_; |
| Call* receiver_call_; |
| }; |
| |
| TEST_F(TransportFeedbackEndToEndTest, VideoReceivesTransportFeedback) { |
| TransportFeedbackTester test(true, 1, 0); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(TransportFeedbackEndToEndTest, VideoTransportFeedbackNotConfigured) { |
| TransportFeedbackTester test(false, 1, 0); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(TransportFeedbackEndToEndTest, AudioReceivesTransportFeedback) { |
| test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); |
| TransportFeedbackTester test(true, 0, 1); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(TransportFeedbackEndToEndTest, AudioTransportFeedbackNotConfigured) { |
| TransportFeedbackTester test(false, 0, 1); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(TransportFeedbackEndToEndTest, AudioVideoReceivesTransportFeedback) { |
| TransportFeedbackTester test(true, 1, 1); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(TransportFeedbackEndToEndTest, |
| StopsAndResumesMediaWhenCongestionWindowFull) { |
| test::ScopedFieldTrials override_field_trials( |
| "WebRTC-CongestionWindow/QueueSize:250/"); |
| |
| class TransportFeedbackTester : public test::EndToEndTest { |
| public: |
| TransportFeedbackTester(size_t num_video_streams, size_t num_audio_streams) |
| : EndToEndTest( |
| ::webrtc::TransportFeedbackEndToEndTest::kDefaultTimeoutMs), |
| num_video_streams_(num_video_streams), |
| num_audio_streams_(num_audio_streams), |
| media_sent_(0), |
| media_sent_before_(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_); |
| // Padding is expected in congested state to probe for connectivity when |
| // packets has been dropped. |
| if (only_padding) { |
| media_sent_before_ = media_sent_; |
| ++padding_sent_; |
| } else { |
| ++media_sent_; |
| if (padding_sent_ == 0) { |
| ++media_sent_before_; |
| EXPECT_LT(media_sent_, 40) |
| << "Media sent without feedback when congestion window is full."; |
| } else if (media_sent_ > media_sent_before_) { |
| observation_complete_.Set(); |
| } |
| } |
| return SEND_PACKET; |
| } |
| |
| Action OnReceiveRtcp(const uint8_t* data, size_t length) override { |
| rtc::CritScope lock(&crit_); |
| // To fill up the congestion window we drop feedback on packets after 20 |
| // packets have been sent. This means that any packets that has not yet |
| // received feedback after that will be considered as oustanding data and |
| // therefore filling up the congestion window. In the congested state, the |
| // pacer should send padding packets to trigger feedback in case all |
| // feedback of previous traffic was lost. This test listens for the |
| // padding packets and when 2 padding packets have been received, feedback |
| // will be let trough again. This should cause the pacer to continue |
| // sending meadia yet again. |
| if (media_sent_ > 20 && HasTransportFeedback(data, length) && |
| padding_sent_ < 2) { |
| 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; |
| } |
| void ModifySenderBitrateConfig( |
| BitrateConstraints* bitrate_config) override { |
| bitrate_config->max_bitrate_bps = 300000; |
| } |
| |
| void PerformTest() override { |
| const int64_t kFailureTimeoutMs = 10000; |
| EXPECT_TRUE(observation_complete_.Wait(kFailureTimeoutMs)) |
| << "Stream not continued after congestion window full."; |
| } |
| |
| size_t GetNumVideoStreams() const override { return num_video_streams_; } |
| size_t GetNumAudioStreams() const override { return num_audio_streams_; } |
| |
| private: |
| const size_t num_video_streams_; |
| const size_t num_audio_streams_; |
| rtc::CriticalSection crit_; |
| int media_sent_ RTC_GUARDED_BY(crit_); |
| int media_sent_before_ RTC_GUARDED_BY(crit_); |
| int padding_sent_ RTC_GUARDED_BY(crit_); |
| } test(1, 0); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(TransportFeedbackEndToEndTest, TransportSeqNumOnAudioAndVideo) { |
| test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/"); |
| static constexpr size_t kMinPacketsToWaitFor = 50; |
| class TransportSequenceNumberTest : public test::EndToEndTest { |
| public: |
| TransportSequenceNumberTest() |
| : EndToEndTest(kDefaultTimeoutMs), |
| video_observed_(false), |
| audio_observed_(false) { |
| parser_->RegisterRtpHeaderExtension(kRtpExtensionTransportSequenceNumber, |
| kTransportSequenceNumberExtensionId); |
| } |
| |
| size_t GetNumVideoStreams() const override { return 1; } |
| size_t GetNumAudioStreams() const override { return 1; } |
| |
| void ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) override { |
| send_config->rtp.extensions.clear(); |
| send_config->rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kTransportSequenceNumberUri, |
| kTransportSequenceNumberExtensionId)); |
| (*receive_configs)[0].rtp.extensions.clear(); |
| (*receive_configs)[0].rtp.extensions = send_config->rtp.extensions; |
| } |
| |
| Action OnSendRtp(const uint8_t* packet, size_t length) override { |
| RTPHeader header; |
| EXPECT_TRUE(parser_->Parse(packet, length, &header)); |
| EXPECT_TRUE(header.extension.hasTransportSequenceNumber); |
| // Unwrap packet id and verify uniqueness. |
| int64_t packet_id = |
| unwrapper_.Unwrap(header.extension.transportSequenceNumber); |
| EXPECT_TRUE(received_packet_ids_.insert(packet_id).second); |
| |
| if (header.ssrc == kVideoSendSsrcs[0]) |
| video_observed_ = true; |
| if (header.ssrc == kAudioSendSsrc) |
| audio_observed_ = true; |
| if (audio_observed_ && video_observed_ && |
| received_packet_ids_.size() >= kMinPacketsToWaitFor) { |
| size_t packet_id_range = |
| *received_packet_ids_.rbegin() - *received_packet_ids_.begin() + 1; |
| EXPECT_EQ(received_packet_ids_.size(), packet_id_range); |
| observation_complete_.Set(); |
| } |
| return SEND_PACKET; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) << "Timed out while waiting for audio and video " |
| "packets with transport sequence number."; |
| } |
| |
| void ExpectSuccessful() { |
| EXPECT_TRUE(video_observed_); |
| EXPECT_TRUE(audio_observed_); |
| EXPECT_GE(received_packet_ids_.size(), kMinPacketsToWaitFor); |
| } |
| |
| private: |
| bool video_observed_; |
| bool audio_observed_; |
| SequenceNumberUnwrapper unwrapper_; |
| std::set<int64_t> received_packet_ids_; |
| } test; |
| |
| RunBaseTest(&test); |
| // Double check conditions for successful test to produce better error |
| // message when the test fail. |
| test.ExpectSuccessful(); |
| } |
| } // namespace webrtc |