/* * 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 "modules/rtp_rtcp/source/rtp_rtcp_impl2.h" #include #include #include #include "absl/types/optional.h" #include "api/transport/field_trial_based_config.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet.h" #include "modules/rtp_rtcp/source/rtcp_packet/nack.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_rtcp_interface.h" #include "modules/rtp_rtcp/source/rtp_sender_video.h" #include "rtc_base/rate_limiter.h" #include "rtc_base/strings/string_builder.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/rtcp_packet_parser.h" #include "test/rtp_header_parser.h" #include "test/run_loop.h" #include "test/time_controller/simulated_time_controller.h" using ::testing::ElementsAre; using ::testing::Eq; using ::testing::Field; using ::testing::Gt; using ::testing::Not; using ::testing::Optional; using ::testing::SizeIs; namespace webrtc { namespace { const uint32_t kSenderSsrc = 0x12345; const uint32_t kReceiverSsrc = 0x23456; const int64_t kOneWayNetworkDelayMs = 100; const uint8_t kBaseLayerTid = 0; const uint8_t kHigherLayerTid = 1; const uint16_t kSequenceNumber = 100; const uint8_t kPayloadType = 100; const int kWidth = 320; const int kHeight = 100; class RtcpRttStatsTestImpl : public RtcpRttStats { public: RtcpRttStatsTestImpl() : rtt_ms_(0) {} ~RtcpRttStatsTestImpl() override = default; void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; } int64_t LastProcessedRtt() const override { return rtt_ms_; } int64_t rtt_ms_; }; class SendTransport : public Transport { public: SendTransport() : receiver_(nullptr), time_controller_(nullptr), delay_ms_(0), rtp_packets_sent_(0), rtcp_packets_sent_(0) {} void SetRtpRtcpModule(ModuleRtpRtcpImpl2* receiver) { receiver_ = receiver; } void SimulateNetworkDelay(int64_t delay_ms, TimeController* time_controller) { time_controller_ = time_controller; delay_ms_ = delay_ms; } bool SendRtp(const uint8_t* data, size_t len, const PacketOptions& options) override { RTPHeader header; std::unique_ptr parser(RtpHeaderParser::CreateForTest()); EXPECT_TRUE(parser->Parse(static_cast(data), len, &header)); ++rtp_packets_sent_; last_rtp_header_ = header; return true; } bool SendRtcp(const uint8_t* data, size_t len) override { test::RtcpPacketParser parser; parser.Parse(data, len); last_nack_list_ = parser.nack()->packet_ids(); if (time_controller_) { time_controller_->AdvanceTime(TimeDelta::Millis(delay_ms_)); } EXPECT_TRUE(receiver_); receiver_->IncomingRtcpPacket(data, len); ++rtcp_packets_sent_; return true; } size_t NumRtcpSent() { return rtcp_packets_sent_; } ModuleRtpRtcpImpl2* receiver_; TimeController* time_controller_; int64_t delay_ms_; int rtp_packets_sent_; size_t rtcp_packets_sent_; RTPHeader last_rtp_header_; std::vector last_nack_list_; }; struct TestConfig { explicit TestConfig(bool with_overhead) : with_overhead(with_overhead) {} bool with_overhead = false; }; class FieldTrialConfig : public WebRtcKeyValueConfig { public: static FieldTrialConfig GetFromTestConfig(const TestConfig& config) { FieldTrialConfig trials; trials.overhead_enabled_ = config.with_overhead; return trials; } FieldTrialConfig() : overhead_enabled_(false), max_padding_factor_(1200) {} ~FieldTrialConfig() override {} void SetOverHeadEnabled(bool enabled) { overhead_enabled_ = enabled; } void SetMaxPaddingFactor(double factor) { max_padding_factor_ = factor; } std::string Lookup(absl::string_view key) const override { if (key == "WebRTC-LimitPaddingSize") { char string_buf[32]; rtc::SimpleStringBuilder ssb(string_buf); ssb << "factor:" << max_padding_factor_; return ssb.str(); } else if (key == "WebRTC-SendSideBwe-WithOverhead") { return overhead_enabled_ ? "Enabled" : "Disabled"; } return ""; } private: bool overhead_enabled_; double max_padding_factor_; }; class RtpRtcpModule : public RtcpPacketTypeCounterObserver { public: RtpRtcpModule(TimeController* time_controller, bool is_sender, const FieldTrialConfig& trials) : is_sender_(is_sender), trials_(trials), receive_statistics_( ReceiveStatistics::Create(time_controller->GetClock())), time_controller_(time_controller) { CreateModuleImpl(); transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, time_controller); } const bool is_sender_; const FieldTrialConfig& trials_; RtcpPacketTypeCounter packets_sent_; RtcpPacketTypeCounter packets_received_; std::unique_ptr receive_statistics_; SendTransport transport_; RtcpRttStatsTestImpl rtt_stats_; std::unique_ptr impl_; int rtcp_report_interval_ms_ = 0; void RtcpPacketTypesCounterUpdated( uint32_t ssrc, const RtcpPacketTypeCounter& packet_counter) override { counter_map_[ssrc] = packet_counter; } RtcpPacketTypeCounter RtcpSent() { // RTCP counters for remote SSRC. return counter_map_[is_sender_ ? kReceiverSsrc : kSenderSsrc]; } RtcpPacketTypeCounter RtcpReceived() { // Received RTCP stats for (own) local SSRC. return counter_map_[impl_->SSRC()]; } int RtpSent() { return transport_.rtp_packets_sent_; } uint16_t LastRtpSequenceNumber() { return transport_.last_rtp_header_.sequenceNumber; } std::vector LastNackListSent() { return transport_.last_nack_list_; } void SetRtcpReportIntervalAndReset(int rtcp_report_interval_ms) { rtcp_report_interval_ms_ = rtcp_report_interval_ms; CreateModuleImpl(); } private: void CreateModuleImpl() { RtpRtcpInterface::Configuration config; config.audio = false; config.clock = time_controller_->GetClock(); config.outgoing_transport = &transport_; config.receive_statistics = receive_statistics_.get(); config.rtcp_packet_type_counter_observer = this; config.rtt_stats = &rtt_stats_; config.rtcp_report_interval_ms = rtcp_report_interval_ms_; config.local_media_ssrc = is_sender_ ? kSenderSsrc : kReceiverSsrc; config.need_rtp_packet_infos = true; config.non_sender_rtt_measurement = true; config.field_trials = &trials_; impl_.reset(new ModuleRtpRtcpImpl2(config)); impl_->SetRemoteSSRC(is_sender_ ? kReceiverSsrc : kSenderSsrc); impl_->SetRTCPStatus(RtcpMode::kCompound); } TimeController* const time_controller_; std::map counter_map_; }; } // namespace class RtpRtcpImpl2Test : public ::testing::TestWithParam { protected: RtpRtcpImpl2Test() : time_controller_(Timestamp::Micros(133590000000000)), field_trials_(FieldTrialConfig::GetFromTestConfig(GetParam())), sender_(&time_controller_, /*is_sender=*/true, field_trials_), receiver_(&time_controller_, /*is_sender=*/false, field_trials_) {} void SetUp() override { // Send module. EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true)); sender_.impl_->SetSendingMediaStatus(true); sender_.impl_->SetSequenceNumber(kSequenceNumber); sender_.impl_->SetStorePacketsStatus(true, 100); RTPSenderVideo::Config video_config; video_config.clock = time_controller_.GetClock(); video_config.rtp_sender = sender_.impl_->RtpSender(); video_config.field_trials = &field_trials_; sender_video_ = std::make_unique(video_config); // Receive module. EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false)); receiver_.impl_->SetSendingMediaStatus(false); // Transport settings. sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get()); receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get()); } void AdvanceTimeMs(int64_t milliseconds) { time_controller_.AdvanceTime(TimeDelta::Millis(milliseconds)); } GlobalSimulatedTimeController time_controller_; FieldTrialConfig field_trials_; RtpRtcpModule sender_; std::unique_ptr sender_video_; RtpRtcpModule receiver_; bool SendFrame(const RtpRtcpModule* module, RTPSenderVideo* sender, uint8_t tid, uint32_t rtp_timestamp) { RTPVideoHeaderVP8 vp8_header = {}; vp8_header.temporalIdx = tid; RTPVideoHeader rtp_video_header; rtp_video_header.frame_type = VideoFrameType::kVideoFrameKey; rtp_video_header.width = kWidth; rtp_video_header.height = kHeight; rtp_video_header.rotation = kVideoRotation_0; rtp_video_header.content_type = VideoContentType::UNSPECIFIED; rtp_video_header.playout_delay = {-1, -1}; rtp_video_header.is_first_packet_in_frame = true; rtp_video_header.simulcastIdx = 0; rtp_video_header.codec = kVideoCodecVP8; rtp_video_header.video_type_header = vp8_header; rtp_video_header.video_timing = {0u, 0u, 0u, 0u, 0u, 0u, false}; const uint8_t payload[100] = {0}; bool success = module->impl_->OnSendingRtpFrame(0, 0, kPayloadType, true); success &= sender->SendVideo(kPayloadType, VideoCodecType::kVideoCodecVP8, rtp_timestamp, 0, payload, rtp_video_header, 0); return success; } void IncomingRtcpNack(const RtpRtcpModule* module, uint16_t sequence_number) { bool sender = module->impl_->SSRC() == kSenderSsrc; rtcp::Nack nack; uint16_t list[1]; list[0] = sequence_number; const uint16_t kListLength = sizeof(list) / sizeof(list[0]); nack.SetSenderSsrc(sender ? kReceiverSsrc : kSenderSsrc); nack.SetMediaSsrc(sender ? kSenderSsrc : kReceiverSsrc); nack.SetPacketIds(list, kListLength); rtc::Buffer packet = nack.Build(); module->impl_->IncomingRtcpPacket(packet.data(), packet.size()); } }; TEST_P(RtpRtcpImpl2Test, RetransmitsAllLayers) { // Send frames. EXPECT_EQ(0, sender_.RtpSent()); EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // kSequenceNumber EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kHigherLayerTid, /*timestamp=*/0)); // kSequenceNumber + 1 EXPECT_TRUE(SendFrame(&sender_, sender_video_.get(), kNoTemporalIdx, /*timestamp=*/0)); // kSequenceNumber + 2 EXPECT_EQ(3, sender_.RtpSent()); EXPECT_EQ(kSequenceNumber + 2, sender_.LastRtpSequenceNumber()); // Min required delay until retransmit = 5 + RTT ms (RTT = 0). AdvanceTimeMs(5); // Frame with kBaseLayerTid re-sent. IncomingRtcpNack(&sender_, kSequenceNumber); EXPECT_EQ(4, sender_.RtpSent()); EXPECT_EQ(kSequenceNumber, sender_.LastRtpSequenceNumber()); // Frame with kHigherLayerTid re-sent. IncomingRtcpNack(&sender_, kSequenceNumber + 1); EXPECT_EQ(5, sender_.RtpSent()); EXPECT_EQ(kSequenceNumber + 1, sender_.LastRtpSequenceNumber()); // Frame with kNoTemporalIdx re-sent. IncomingRtcpNack(&sender_, kSequenceNumber + 2); EXPECT_EQ(6, sender_.RtpSent()); EXPECT_EQ(kSequenceNumber + 2, sender_.LastRtpSequenceNumber()); } TEST_P(RtpRtcpImpl2Test, Rtt) { RtpPacketReceived packet; packet.SetTimestamp(1); packet.SetSequenceNumber(123); packet.SetSsrc(kSenderSsrc); packet.AllocatePayload(100 - 12); receiver_.receive_statistics_->OnRtpPacket(packet); // Send Frame before sending an SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Sender module should send an SR. EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport)); // Receiver module should send a RR with a response to the last received SR. AdvanceTimeMs(1000); EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport)); // Verify RTT. int64_t rtt; int64_t avg_rtt; int64_t min_rtt; int64_t max_rtt; EXPECT_EQ( 0, sender_.impl_->RTT(kReceiverSsrc, &rtt, &avg_rtt, &min_rtt, &max_rtt)); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, rtt, 1); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, avg_rtt, 1); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, min_rtt, 1); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, max_rtt, 1); // No RTT from other ssrc. EXPECT_EQ(-1, sender_.impl_->RTT(kReceiverSsrc + 1, &rtt, &avg_rtt, &min_rtt, &max_rtt)); // Verify RTT from rtt_stats config. EXPECT_EQ(0, sender_.rtt_stats_.LastProcessedRtt()); EXPECT_EQ(0, sender_.impl_->rtt_ms()); AdvanceTimeMs(1000); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, sender_.rtt_stats_.LastProcessedRtt(), 1); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, sender_.impl_->rtt_ms(), 1); } TEST_P(RtpRtcpImpl2Test, RttForReceiverOnly) { // Receiver module should send a Receiver time reference report (RTRR). EXPECT_EQ(0, receiver_.impl_->SendRTCP(kRtcpReport)); // Sender module should send a response to the last received RTRR (DLRR). AdvanceTimeMs(1000); // Send Frame before sending a SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); EXPECT_EQ(0, sender_.impl_->SendRTCP(kRtcpReport)); // Verify RTT. EXPECT_EQ(0, receiver_.rtt_stats_.LastProcessedRtt()); EXPECT_EQ(0, receiver_.impl_->rtt_ms()); AdvanceTimeMs(1000); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, receiver_.rtt_stats_.LastProcessedRtt(), 1); EXPECT_NEAR(2 * kOneWayNetworkDelayMs, receiver_.impl_->rtt_ms(), 1); } TEST_P(RtpRtcpImpl2Test, NoSrBeforeMedia) { // Ignore fake transport delays in this test. sender_.transport_.SimulateNetworkDelay(0, &time_controller_); receiver_.transport_.SimulateNetworkDelay(0, &time_controller_); sender_.impl_->Process(); EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms); // Verify no SR is sent before media has been sent, RR should still be sent // from the receiving module though. AdvanceTimeMs(2000); int64_t current_time = time_controller_.GetClock()->TimeInMilliseconds(); sender_.impl_->Process(); receiver_.impl_->Process(); EXPECT_EQ(-1, sender_.RtcpSent().first_packet_time_ms); EXPECT_EQ(receiver_.RtcpSent().first_packet_time_ms, current_time); EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, current_time); } TEST_P(RtpRtcpImpl2Test, RtcpPacketTypeCounter_Nack) { EXPECT_EQ(-1, receiver_.RtcpSent().first_packet_time_ms); EXPECT_EQ(-1, sender_.RtcpReceived().first_packet_time_ms); EXPECT_EQ(0U, sender_.RtcpReceived().nack_packets); EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets); // Receive module sends a NACK. const uint16_t kNackLength = 1; uint16_t nack_list[kNackLength] = {123}; EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets); EXPECT_GT(receiver_.RtcpSent().first_packet_time_ms, -1); // Send module receives the NACK. EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets); EXPECT_GT(sender_.RtcpReceived().first_packet_time_ms, -1); } TEST_P(RtpRtcpImpl2Test, AddStreamDataCounters) { StreamDataCounters rtp; const int64_t kStartTimeMs = 1; rtp.first_packet_time_ms = kStartTimeMs; rtp.transmitted.packets = 1; rtp.transmitted.payload_bytes = 1; rtp.transmitted.header_bytes = 2; rtp.transmitted.padding_bytes = 3; EXPECT_EQ(rtp.transmitted.TotalBytes(), rtp.transmitted.payload_bytes + rtp.transmitted.header_bytes + rtp.transmitted.padding_bytes); StreamDataCounters rtp2; rtp2.first_packet_time_ms = -1; rtp2.transmitted.packets = 10; rtp2.transmitted.payload_bytes = 10; rtp2.retransmitted.header_bytes = 4; rtp2.retransmitted.payload_bytes = 5; rtp2.retransmitted.padding_bytes = 6; rtp2.retransmitted.packets = 7; rtp2.fec.packets = 8; StreamDataCounters sum = rtp; sum.Add(rtp2); EXPECT_EQ(kStartTimeMs, sum.first_packet_time_ms); EXPECT_EQ(11U, sum.transmitted.packets); EXPECT_EQ(11U, sum.transmitted.payload_bytes); EXPECT_EQ(2U, sum.transmitted.header_bytes); EXPECT_EQ(3U, sum.transmitted.padding_bytes); EXPECT_EQ(4U, sum.retransmitted.header_bytes); EXPECT_EQ(5U, sum.retransmitted.payload_bytes); EXPECT_EQ(6U, sum.retransmitted.padding_bytes); EXPECT_EQ(7U, sum.retransmitted.packets); EXPECT_EQ(8U, sum.fec.packets); EXPECT_EQ(sum.transmitted.TotalBytes(), rtp.transmitted.TotalBytes() + rtp2.transmitted.TotalBytes()); StreamDataCounters rtp3; rtp3.first_packet_time_ms = kStartTimeMs + 10; sum.Add(rtp3); EXPECT_EQ(kStartTimeMs, sum.first_packet_time_ms); // Holds oldest time. } TEST_P(RtpRtcpImpl2Test, SendsInitialNackList) { // Send module sends a NACK. const uint16_t kNackLength = 1; uint16_t nack_list[kNackLength] = {123}; EXPECT_EQ(0U, sender_.RtcpSent().nack_packets); // Send Frame before sending a compound RTCP that starts with SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, sender_.RtcpSent().nack_packets); EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123)); } TEST_P(RtpRtcpImpl2Test, SendsExtendedNackList) { // Send module sends a NACK. const uint16_t kNackLength = 1; uint16_t nack_list[kNackLength] = {123}; EXPECT_EQ(0U, sender_.RtcpSent().nack_packets); // Send Frame before sending a compound RTCP that starts with SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, sender_.RtcpSent().nack_packets); EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123)); // Same list not re-send. EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, sender_.RtcpSent().nack_packets); EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123)); // Only extended list sent. const uint16_t kNackExtLength = 2; uint16_t nack_list_ext[kNackExtLength] = {123, 124}; EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list_ext, kNackExtLength)); EXPECT_EQ(2U, sender_.RtcpSent().nack_packets); EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(124)); } TEST_P(RtpRtcpImpl2Test, ReSendsNackListAfterRttMs) { sender_.transport_.SimulateNetworkDelay(0, &time_controller_); // Send module sends a NACK. const uint16_t kNackLength = 2; uint16_t nack_list[kNackLength] = {123, 125}; EXPECT_EQ(0U, sender_.RtcpSent().nack_packets); // Send Frame before sending a compound RTCP that starts with SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, sender_.RtcpSent().nack_packets); EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123, 125)); // Same list not re-send, rtt interval has not passed. const int kStartupRttMs = 100; AdvanceTimeMs(kStartupRttMs); EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, sender_.RtcpSent().nack_packets); // Rtt interval passed, full list sent. AdvanceTimeMs(1); EXPECT_EQ(0, sender_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(2U, sender_.RtcpSent().nack_packets); EXPECT_THAT(sender_.LastNackListSent(), ElementsAre(123, 125)); } TEST_P(RtpRtcpImpl2Test, UniqueNackRequests) { receiver_.transport_.SimulateNetworkDelay(0, &time_controller_); EXPECT_EQ(0U, receiver_.RtcpSent().nack_packets); EXPECT_EQ(0U, receiver_.RtcpSent().nack_requests); EXPECT_EQ(0U, receiver_.RtcpSent().unique_nack_requests); EXPECT_EQ(0, receiver_.RtcpSent().UniqueNackRequestsInPercent()); // Receive module sends NACK request. const uint16_t kNackLength = 4; uint16_t nack_list[kNackLength] = {10, 11, 13, 18}; EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list, kNackLength)); EXPECT_EQ(1U, receiver_.RtcpSent().nack_packets); EXPECT_EQ(4U, receiver_.RtcpSent().nack_requests); EXPECT_EQ(4U, receiver_.RtcpSent().unique_nack_requests); EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(10, 11, 13, 18)); // Send module receives the request. EXPECT_EQ(1U, sender_.RtcpReceived().nack_packets); EXPECT_EQ(4U, sender_.RtcpReceived().nack_requests); EXPECT_EQ(4U, sender_.RtcpReceived().unique_nack_requests); EXPECT_EQ(100, sender_.RtcpReceived().UniqueNackRequestsInPercent()); // Receive module sends new request with duplicated packets. const int kStartupRttMs = 100; AdvanceTimeMs(kStartupRttMs + 1); const uint16_t kNackLength2 = 4; uint16_t nack_list2[kNackLength2] = {11, 18, 20, 21}; EXPECT_EQ(0, receiver_.impl_->SendNACK(nack_list2, kNackLength2)); EXPECT_EQ(2U, receiver_.RtcpSent().nack_packets); EXPECT_EQ(8U, receiver_.RtcpSent().nack_requests); EXPECT_EQ(6U, receiver_.RtcpSent().unique_nack_requests); EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(11, 18, 20, 21)); // Send module receives the request. EXPECT_EQ(2U, sender_.RtcpReceived().nack_packets); EXPECT_EQ(8U, sender_.RtcpReceived().nack_requests); EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests); EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent()); } TEST_P(RtpRtcpImpl2Test, ConfigurableRtcpReportInterval) { const int kVideoReportInterval = 3000; // Recreate sender impl with new configuration, and redo setup. sender_.SetRtcpReportIntervalAndReset(kVideoReportInterval); SetUp(); EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Initial state sender_.impl_->Process(); EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, -1); EXPECT_EQ(0u, sender_.transport_.NumRtcpSent()); // Move ahead to the last ms before a rtcp is expected, no action. AdvanceTimeMs(kVideoReportInterval / 2 - 1); sender_.impl_->Process(); EXPECT_EQ(sender_.RtcpSent().first_packet_time_ms, -1); EXPECT_EQ(sender_.transport_.NumRtcpSent(), 0u); // Move ahead to the first rtcp. Send RTCP. AdvanceTimeMs(1); sender_.impl_->Process(); EXPECT_GT(sender_.RtcpSent().first_packet_time_ms, -1); EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u); EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Move ahead to the last possible second before second rtcp is expected. AdvanceTimeMs(kVideoReportInterval * 1 / 2 - 1); sender_.impl_->Process(); EXPECT_EQ(sender_.transport_.NumRtcpSent(), 1u); // Move ahead into the range of second rtcp, the second rtcp may be sent. AdvanceTimeMs(1); sender_.impl_->Process(); EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u); AdvanceTimeMs(kVideoReportInterval / 2); sender_.impl_->Process(); EXPECT_GE(sender_.transport_.NumRtcpSent(), 1u); // Move out the range of second rtcp, the second rtcp must have been sent. AdvanceTimeMs(kVideoReportInterval / 2); sender_.impl_->Process(); EXPECT_EQ(sender_.transport_.NumRtcpSent(), 2u); } TEST_P(RtpRtcpImpl2Test, StoresPacketInfoForSentPackets) { const uint32_t kStartTimestamp = 1u; SetUp(); sender_.impl_->SetStartTimestamp(kStartTimestamp); sender_.impl_->SetSequenceNumber(1); PacedPacketInfo pacing_info; RtpPacketToSend packet(nullptr); packet.set_packet_type(RtpPacketToSend::Type::kVideo); packet.SetSsrc(kSenderSsrc); // Single-packet frame. packet.SetTimestamp(1); packet.SetSequenceNumber(1); packet.set_first_packet_of_frame(true); packet.SetMarker(true); sender_.impl_->TrySendPacket(&packet, pacing_info); AdvanceTimeMs(1); std::vector seqno_info = sender_.impl_->GetSentRtpPacketInfos(std::vector{1}); EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info( /*timestamp=*/1 - kStartTimestamp, /*is_first=*/1, /*is_last=*/1))); // Three-packet frame. packet.SetTimestamp(2); packet.SetSequenceNumber(2); packet.set_first_packet_of_frame(true); packet.SetMarker(false); sender_.impl_->TrySendPacket(&packet, pacing_info); packet.SetSequenceNumber(3); packet.set_first_packet_of_frame(false); sender_.impl_->TrySendPacket(&packet, pacing_info); packet.SetSequenceNumber(4); packet.SetMarker(true); sender_.impl_->TrySendPacket(&packet, pacing_info); AdvanceTimeMs(1); seqno_info = sender_.impl_->GetSentRtpPacketInfos(std::vector{2, 3, 4}); EXPECT_THAT(seqno_info, ElementsAre(RtpSequenceNumberMap::Info( /*timestamp=*/2 - kStartTimestamp, /*is_first=*/1, /*is_last=*/0), RtpSequenceNumberMap::Info( /*timestamp=*/2 - kStartTimestamp, /*is_first=*/0, /*is_last=*/0), RtpSequenceNumberMap::Info( /*timestamp=*/2 - kStartTimestamp, /*is_first=*/0, /*is_last=*/1))); } // Checks that the sender report stats are not available if no RTCP SR was sent. TEST_P(RtpRtcpImpl2Test, SenderReportStatsNotAvailable) { EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt)); } // Checks that the sender report stats are available if an RTCP SR was sent. TEST_P(RtpRtcpImpl2Test, SenderReportStatsAvailable) { // Send a frame in order to send an SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Send an SR. ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0)); EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Not(Eq(absl::nullopt))); } // Checks that the sender report stats are not available if an RTCP SR with an // unexpected SSRC is received. TEST_P(RtpRtcpImpl2Test, SenderReportStatsNotUpdatedWithUnexpectedSsrc) { constexpr uint32_t kUnexpectedSenderSsrc = 0x87654321; static_assert(kUnexpectedSenderSsrc != kSenderSsrc, ""); // Forge a sender report and pass it to the receiver as if an RTCP SR were // sent by an unexpected sender. rtcp::SenderReport sr; sr.SetSenderSsrc(kUnexpectedSenderSsrc); sr.SetNtp({/*seconds=*/1u, /*fractions=*/1u << 31}); sr.SetPacketCount(123u); sr.SetOctetCount(456u); auto raw_packet = sr.Build(); receiver_.impl_->IncomingRtcpPacket(raw_packet.data(), raw_packet.size()); EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Eq(absl::nullopt)); } // Checks the stats derived from the last received RTCP SR are set correctly. TEST_P(RtpRtcpImpl2Test, SenderReportStatsCheckStatsFromLastReport) { using SenderReportStats = RtpRtcpInterface::SenderReportStats; const NtpTime ntp(/*seconds=*/1u, /*fractions=*/1u << 31); constexpr uint32_t kPacketCount = 123u; constexpr uint32_t kOctetCount = 456u; // Forge a sender report and pass it to the receiver as if an RTCP SR were // sent by the sender. rtcp::SenderReport sr; sr.SetSenderSsrc(kSenderSsrc); sr.SetNtp(ntp); sr.SetPacketCount(kPacketCount); sr.SetOctetCount(kOctetCount); auto raw_packet = sr.Build(); receiver_.impl_->IncomingRtcpPacket(raw_packet.data(), raw_packet.size()); EXPECT_THAT( receiver_.impl_->GetSenderReportStats(), Optional(AllOf(Field(&SenderReportStats::last_remote_timestamp, Eq(ntp)), Field(&SenderReportStats::packets_sent, Eq(kPacketCount)), Field(&SenderReportStats::bytes_sent, Eq(kOctetCount))))); } // Checks that the sender report stats count equals the number of sent RTCP SRs. TEST_P(RtpRtcpImpl2Test, SenderReportStatsCount) { using SenderReportStats = RtpRtcpInterface::SenderReportStats; // Send a frame in order to send an SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Send the first SR. ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0)); EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Optional(Field(&SenderReportStats::reports_count, Eq(1u)))); // Send the second SR. ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0)); EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Optional(Field(&SenderReportStats::reports_count, Eq(2u)))); } // Checks that the sender report stats include a valid arrival time if an RTCP // SR was sent. TEST_P(RtpRtcpImpl2Test, SenderReportStatsArrivalTimestampSet) { // Send a frame in order to send an SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Send an SR. ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0)); auto stats = receiver_.impl_->GetSenderReportStats(); ASSERT_THAT(stats, Not(Eq(absl::nullopt))); EXPECT_TRUE(stats->last_arrival_timestamp.Valid()); } // Checks that the packet and byte counters from an RTCP SR are not zero once // a frame is sent. TEST_P(RtpRtcpImpl2Test, SenderReportStatsPacketByteCounters) { using SenderReportStats = RtpRtcpInterface::SenderReportStats; // Send a frame in order to send an SR. EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Gt(0)); // Advance time otherwise the RTCP SR report will not include any packets // generated by `SendFrame()`. AdvanceTimeMs(1); // Send an SR. ASSERT_THAT(sender_.impl_->SendRTCP(kRtcpReport), Eq(0)); EXPECT_THAT(receiver_.impl_->GetSenderReportStats(), Optional(AllOf(Field(&SenderReportStats::packets_sent, Gt(0u)), Field(&SenderReportStats::bytes_sent, Gt(0u))))); } TEST_P(RtpRtcpImpl2Test, SendingVideoAdvancesSequenceNumber) { const uint16_t sequence_number = sender_.impl_->SequenceNumber(); EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Gt(0)); EXPECT_EQ(sequence_number + 1, sender_.impl_->SequenceNumber()); } TEST_P(RtpRtcpImpl2Test, SequenceNumberNotAdvancedWhenNotSending) { const uint16_t sequence_number = sender_.impl_->SequenceNumber(); sender_.impl_->SetSendingMediaStatus(false); EXPECT_FALSE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); ASSERT_THAT(sender_.transport_.rtp_packets_sent_, Eq(0)); EXPECT_EQ(sequence_number, sender_.impl_->SequenceNumber()); } TEST_P(RtpRtcpImpl2Test, PaddingNotAllowedInMiddleOfFrame) { constexpr size_t kPaddingSize = 100; // Can't send padding before media. EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(0u)); EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, /*timestamp=*/0)); // Padding is now ok. EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(Gt(0u))); // Send half a video frame. PacedPacketInfo pacing_info; std::unique_ptr packet = sender_.impl_->RtpSender()->AllocatePacket(); packet->set_packet_type(RtpPacketToSend::Type::kVideo); packet->set_first_packet_of_frame(true); packet->SetMarker(false); // Marker false - not last packet of frame. sender_.impl_->RtpSender()->AssignSequenceNumber(packet.get()); EXPECT_TRUE(sender_.impl_->TrySendPacket(packet.get(), pacing_info)); // Padding not allowed in middle of frame. EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(0u)); packet = sender_.impl_->RtpSender()->AllocatePacket(); packet->set_packet_type(RtpPacketToSend::Type::kVideo); packet->set_first_packet_of_frame(true); packet->SetMarker(true); sender_.impl_->RtpSender()->AssignSequenceNumber(packet.get()); EXPECT_TRUE(sender_.impl_->TrySendPacket(packet.get(), pacing_info)); // Padding is OK again. EXPECT_THAT(sender_.impl_->GeneratePadding(kPaddingSize), SizeIs(Gt(0u))); } TEST_P(RtpRtcpImpl2Test, PaddingTimestampMatchesMedia) { constexpr size_t kPaddingSize = 100; uint32_t kTimestamp = 123; EXPECT_TRUE( SendFrame(&sender_, sender_video_.get(), kBaseLayerTid, kTimestamp)); EXPECT_EQ(sender_.transport_.last_rtp_header_.timestamp, kTimestamp); uint16_t media_seq = sender_.transport_.last_rtp_header_.sequenceNumber; // Generate and send padding. auto padding = sender_.impl_->GeneratePadding(kPaddingSize); ASSERT_FALSE(padding.empty()); for (auto& packet : padding) { sender_.impl_->TrySendPacket(packet.get(), PacedPacketInfo()); } // Verify we sent a new packet, but with the same timestamp. EXPECT_NE(sender_.transport_.last_rtp_header_.sequenceNumber, media_seq); EXPECT_EQ(sender_.transport_.last_rtp_header_.timestamp, kTimestamp); } INSTANTIATE_TEST_SUITE_P(WithAndWithoutOverhead, RtpRtcpImpl2Test, ::testing::Values(TestConfig{false}, TestConfig{true})); } // namespace webrtc