| /* |
| * 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 "testing/gtest/include/gtest/gtest.h" |
| #include "webrtc/modules/rtp_rtcp/interface/receive_statistics.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_header_parser.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_payload_registry.h" |
| #include "webrtc/modules/rtp_rtcp/interface/rtp_rtcp.h" |
| #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h" |
| #include "webrtc/system_wrappers/interface/critical_section_wrapper.h" |
| #include "webrtc/test/testsupport/perf_test.h" |
| #include "webrtc/video/rampup_tests.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| static const int kMaxPacketSize = 1500; |
| |
| std::vector<uint32_t> GenerateSsrcs(size_t num_streams, |
| uint32_t ssrc_offset) { |
| std::vector<uint32_t> ssrcs; |
| for (size_t i = 0; i != num_streams; ++i) |
| ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i)); |
| return ssrcs; |
| } |
| } // namespace |
| |
| StreamObserver::StreamObserver(const SsrcMap& rtx_media_ssrcs, |
| newapi::Transport* feedback_transport, |
| Clock* clock, |
| RemoteBitrateEstimatorFactory* rbe_factory, |
| RateControlType control_type) |
| : clock_(clock), |
| test_done_(EventWrapper::Create()), |
| rtp_parser_(RtpHeaderParser::Create()), |
| feedback_transport_(feedback_transport), |
| receive_stats_(ReceiveStatistics::Create(clock)), |
| payload_registry_( |
| new RTPPayloadRegistry(RTPPayloadStrategy::CreateStrategy(false))), |
| crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| expected_bitrate_bps_(0), |
| start_bitrate_bps_(0), |
| rtx_media_ssrcs_(rtx_media_ssrcs), |
| total_sent_(0), |
| padding_sent_(0), |
| rtx_media_sent_(0), |
| total_packets_sent_(0), |
| padding_packets_sent_(0), |
| rtx_media_packets_sent_(0), |
| test_start_ms_(clock_->TimeInMilliseconds()), |
| ramp_up_finished_ms_(0) { |
| // Ideally we would only have to instantiate an RtcpSender, an |
| // RtpHeaderParser and a RemoteBitrateEstimator here, but due to the current |
| // state of the RTP module we need a full module and receive statistics to |
| // be able to produce an RTCP with REMB. |
| RtpRtcp::Configuration config; |
| config.receive_statistics = receive_stats_.get(); |
| feedback_transport_.Enable(); |
| config.outgoing_transport = &feedback_transport_; |
| rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); |
| rtp_rtcp_->SetREMBStatus(true); |
| rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); |
| rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, |
| kAbsSendTimeExtensionId); |
| rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000; |
| remote_bitrate_estimator_.reset( |
| rbe_factory->Create(this, clock, control_type, |
| kRemoteBitrateEstimatorMinBitrateBps)); |
| } |
| |
| void StreamObserver::set_expected_bitrate_bps( |
| unsigned int expected_bitrate_bps) { |
| CriticalSectionScoped lock(crit_.get()); |
| expected_bitrate_bps_ = expected_bitrate_bps; |
| } |
| |
| void StreamObserver::set_start_bitrate_bps(unsigned int start_bitrate_bps) { |
| CriticalSectionScoped lock(crit_.get()); |
| start_bitrate_bps_ = start_bitrate_bps; |
| } |
| |
| void StreamObserver::OnReceiveBitrateChanged( |
| const std::vector<unsigned int>& ssrcs, unsigned int bitrate) { |
| CriticalSectionScoped lock(crit_.get()); |
| assert(expected_bitrate_bps_ > 0); |
| if (start_bitrate_bps_ != 0) { |
| // For tests with an explicitly set start bitrate, verify the first |
| // bitrate estimate is close to the start bitrate and lower than the |
| // test target bitrate. This is to verify a call respects the configured |
| // start bitrate, but due to the BWE implementation we can't guarantee the |
| // first estimate really is as high as the start bitrate. |
| EXPECT_GT(bitrate, 0.9 * start_bitrate_bps_); |
| EXPECT_LT(bitrate, expected_bitrate_bps_); |
| start_bitrate_bps_ = 0; |
| } |
| if (bitrate >= expected_bitrate_bps_) { |
| ramp_up_finished_ms_ = clock_->TimeInMilliseconds(); |
| // Just trigger if there was any rtx padding packet. |
| if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) { |
| TriggerTestDone(); |
| } |
| } |
| rtp_rtcp_->SetREMBData( |
| bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); |
| rtp_rtcp_->Process(); |
| } |
| |
| bool StreamObserver::SendRtp(const uint8_t* packet, size_t length) { |
| CriticalSectionScoped lock(crit_.get()); |
| RTPHeader header; |
| EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); |
| receive_stats_->IncomingPacket(header, length, false); |
| payload_registry_->SetIncomingPayloadType(header); |
| remote_bitrate_estimator_->IncomingPacket( |
| clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); |
| if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { |
| remote_bitrate_estimator_->Process(); |
| } |
| total_sent_ += length; |
| padding_sent_ += header.paddingLength; |
| ++total_packets_sent_; |
| if (header.paddingLength > 0) |
| ++padding_packets_sent_; |
| if (rtx_media_ssrcs_.find(header.ssrc) != rtx_media_ssrcs_.end()) { |
| rtx_media_sent_ += length - header.headerLength - header.paddingLength; |
| if (header.paddingLength == 0) |
| ++rtx_media_packets_sent_; |
| uint8_t restored_packet[kMaxPacketSize]; |
| uint8_t* restored_packet_ptr = restored_packet; |
| int restored_length = static_cast<int>(length); |
| payload_registry_->RestoreOriginalPacket(&restored_packet_ptr, |
| packet, |
| &restored_length, |
| rtx_media_ssrcs_[header.ssrc], |
| header); |
| length = restored_length; |
| EXPECT_TRUE(rtp_parser_->Parse( |
| restored_packet, static_cast<int>(length), &header)); |
| } else { |
| rtp_rtcp_->SetRemoteSSRC(header.ssrc); |
| } |
| return true; |
| } |
| |
| bool StreamObserver::SendRtcp(const uint8_t* packet, size_t length) { |
| return true; |
| } |
| |
| EventTypeWrapper StreamObserver::Wait() { return test_done_->Wait(120 * 1000); } |
| |
| void StreamObserver::ReportResult(const std::string& measurement, |
| size_t value, |
| const std::string& units) { |
| webrtc::test::PrintResult( |
| measurement, "", |
| ::testing::UnitTest::GetInstance()->current_test_info()->name(), |
| value, units, false); |
| } |
| |
| void StreamObserver::TriggerTestDone() EXCLUSIVE_LOCKS_REQUIRED(crit_) { |
| ReportResult("ramp-up-total-sent", total_sent_, "bytes"); |
| ReportResult("ramp-up-padding-sent", padding_sent_, "bytes"); |
| ReportResult("ramp-up-rtx-media-sent", rtx_media_sent_, "bytes"); |
| ReportResult("ramp-up-total-packets-sent", total_packets_sent_, "packets"); |
| ReportResult("ramp-up-padding-packets-sent", |
| padding_packets_sent_, |
| "packets"); |
| ReportResult("ramp-up-rtx-packets-sent", |
| rtx_media_packets_sent_, |
| "packets"); |
| ReportResult("ramp-up-time", |
| ramp_up_finished_ms_ - test_start_ms_, |
| "milliseconds"); |
| test_done_->Set(); |
| } |
| |
| LowRateStreamObserver::LowRateStreamObserver( |
| newapi::Transport* feedback_transport, |
| Clock* clock, |
| size_t number_of_streams, |
| bool rtx_used) |
| : clock_(clock), |
| number_of_streams_(number_of_streams), |
| rtx_used_(rtx_used), |
| test_done_(EventWrapper::Create()), |
| rtp_parser_(RtpHeaderParser::Create()), |
| feedback_transport_(feedback_transport), |
| receive_stats_(ReceiveStatistics::Create(clock)), |
| crit_(CriticalSectionWrapper::CreateCriticalSection()), |
| send_stream_(NULL), |
| test_state_(kFirstRampup), |
| state_start_ms_(clock_->TimeInMilliseconds()), |
| interval_start_ms_(state_start_ms_), |
| last_remb_bps_(0), |
| sent_bytes_(0), |
| total_overuse_bytes_(0), |
| suspended_in_stats_(false) { |
| RtpRtcp::Configuration config; |
| config.receive_statistics = receive_stats_.get(); |
| feedback_transport_.Enable(); |
| config.outgoing_transport = &feedback_transport_; |
| rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); |
| rtp_rtcp_->SetREMBStatus(true); |
| rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); |
| rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionTransmissionTimeOffset, |
| kTransmissionTimeOffsetExtensionId); |
| AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory; |
| const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 10000; |
| remote_bitrate_estimator_.reset( |
| rbe_factory.Create(this, clock, kMimdControl, |
| kRemoteBitrateEstimatorMinBitrateBps)); |
| forward_transport_config_.link_capacity_kbps = |
| kHighBandwidthLimitBps / 1000; |
| forward_transport_config_.queue_length_packets = 100; // Something large. |
| test::DirectTransport::SetConfig(forward_transport_config_); |
| test::DirectTransport::SetReceiver(this); |
| } |
| |
| void LowRateStreamObserver::SetSendStream(const VideoSendStream* send_stream) { |
| CriticalSectionScoped lock(crit_.get()); |
| send_stream_ = send_stream; |
| } |
| |
| void LowRateStreamObserver::OnReceiveBitrateChanged( |
| const std::vector<unsigned int>& ssrcs, |
| unsigned int bitrate) { |
| CriticalSectionScoped lock(crit_.get()); |
| rtp_rtcp_->SetREMBData( |
| bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); |
| rtp_rtcp_->Process(); |
| last_remb_bps_ = bitrate; |
| } |
| |
| bool LowRateStreamObserver::SendRtp(const uint8_t* data, size_t length) { |
| CriticalSectionScoped lock(crit_.get()); |
| sent_bytes_ += length; |
| int64_t now_ms = clock_->TimeInMilliseconds(); |
| if (now_ms > interval_start_ms_ + 1000) { // Let at least 1 second pass. |
| // Verify that the send rate was about right. |
| unsigned int average_rate_bps = static_cast<unsigned int>(sent_bytes_) * |
| 8 * 1000 / (now_ms - interval_start_ms_); |
| // TODO(holmer): Why is this failing? |
| // EXPECT_LT(average_rate_bps, last_remb_bps_ * 1.1); |
| if (average_rate_bps > last_remb_bps_ * 1.1) { |
| total_overuse_bytes_ += |
| sent_bytes_ - |
| last_remb_bps_ / 8 * (now_ms - interval_start_ms_) / 1000; |
| } |
| EvolveTestState(average_rate_bps); |
| interval_start_ms_ = now_ms; |
| sent_bytes_ = 0; |
| } |
| return test::DirectTransport::SendRtp(data, length); |
| } |
| |
| PacketReceiver::DeliveryStatus LowRateStreamObserver::DeliverPacket( |
| const uint8_t* packet, size_t length) { |
| CriticalSectionScoped lock(crit_.get()); |
| RTPHeader header; |
| EXPECT_TRUE(rtp_parser_->Parse(packet, static_cast<int>(length), &header)); |
| receive_stats_->IncomingPacket(header, length, false); |
| remote_bitrate_estimator_->IncomingPacket( |
| clock_->TimeInMilliseconds(), static_cast<int>(length - 12), header); |
| if (remote_bitrate_estimator_->TimeUntilNextProcess() <= 0) { |
| remote_bitrate_estimator_->Process(); |
| } |
| suspended_in_stats_ = send_stream_->GetStats().suspended; |
| return DELIVERY_OK; |
| } |
| |
| bool LowRateStreamObserver::SendRtcp(const uint8_t* packet, size_t length) { |
| return true; |
| } |
| |
| std::string LowRateStreamObserver::GetModifierString() { |
| std::string str("_"); |
| char temp_str[5]; |
| sprintf(temp_str, "%i", |
| static_cast<int>(number_of_streams_)); |
| str += std::string(temp_str); |
| str += "stream"; |
| str += (number_of_streams_ > 1 ? "s" : ""); |
| str += "_"; |
| str += (rtx_used_ ? "" : "no"); |
| str += "rtx"; |
| return str; |
| } |
| |
| void LowRateStreamObserver::EvolveTestState(unsigned int bitrate_bps) { |
| int64_t now = clock_->TimeInMilliseconds(); |
| CriticalSectionScoped lock(crit_.get()); |
| assert(send_stream_ != NULL); |
| switch (test_state_) { |
| case kFirstRampup: { |
| EXPECT_FALSE(suspended_in_stats_); |
| if (bitrate_bps > kExpectedHighBitrateBps) { |
| // The first ramp-up has reached the target bitrate. Change the |
| // channel limit, and move to the next test state. |
| forward_transport_config_.link_capacity_kbps = |
| kLowBandwidthLimitBps / 1000; |
| test::DirectTransport::SetConfig(forward_transport_config_); |
| test_state_ = kLowRate; |
| webrtc::test::PrintResult("ramp_up_down_up", |
| GetModifierString(), |
| "first_rampup", |
| now - state_start_ms_, |
| "ms", |
| false); |
| state_start_ms_ = now; |
| interval_start_ms_ = now; |
| sent_bytes_ = 0; |
| } |
| break; |
| } |
| case kLowRate: { |
| if (bitrate_bps < kExpectedLowBitrateBps && suspended_in_stats_) { |
| // The ramp-down was successful. Change the channel limit back to a |
| // high value, and move to the next test state. |
| forward_transport_config_.link_capacity_kbps = |
| kHighBandwidthLimitBps / 1000; |
| test::DirectTransport::SetConfig(forward_transport_config_); |
| test_state_ = kSecondRampup; |
| webrtc::test::PrintResult("ramp_up_down_up", |
| GetModifierString(), |
| "rampdown", |
| now - state_start_ms_, |
| "ms", |
| false); |
| state_start_ms_ = now; |
| interval_start_ms_ = now; |
| sent_bytes_ = 0; |
| } |
| break; |
| } |
| case kSecondRampup: { |
| if (bitrate_bps > kExpectedHighBitrateBps && !suspended_in_stats_) { |
| webrtc::test::PrintResult("ramp_up_down_up", |
| GetModifierString(), |
| "second_rampup", |
| now - state_start_ms_, |
| "ms", |
| false); |
| webrtc::test::PrintResult("ramp_up_down_up", |
| GetModifierString(), |
| "total_overuse", |
| total_overuse_bytes_, |
| "bytes", |
| false); |
| test_done_->Set(); |
| } |
| break; |
| } |
| } |
| } |
| |
| EventTypeWrapper LowRateStreamObserver::Wait() { |
| return test_done_->Wait(test::CallTest::kLongTimeoutMs); |
| } |
| |
| void RampUpTest::RunRampUpTest(bool rtx, |
| size_t num_streams, |
| unsigned int start_bitrate_bps, |
| const std::string& extension_type) { |
| std::vector<uint32_t> ssrcs(GenerateSsrcs(num_streams, 100)); |
| std::vector<uint32_t> rtx_ssrcs(GenerateSsrcs(num_streams, 200)); |
| StreamObserver::SsrcMap rtx_ssrc_map; |
| if (rtx) { |
| for (size_t i = 0; i < ssrcs.size(); ++i) |
| rtx_ssrc_map[rtx_ssrcs[i]] = ssrcs[i]; |
| } |
| |
| CreateSendConfig(num_streams); |
| |
| scoped_ptr<RemoteBitrateEstimatorFactory> rbe_factory; |
| RateControlType control_type; |
| if (extension_type == RtpExtension::kAbsSendTime) { |
| control_type = kAimdControl; |
| rbe_factory.reset(new AbsoluteSendTimeRemoteBitrateEstimatorFactory); |
| send_config_.rtp.extensions.push_back(RtpExtension( |
| extension_type.c_str(), kAbsSendTimeExtensionId)); |
| } else { |
| control_type = kMimdControl; |
| rbe_factory.reset(new RemoteBitrateEstimatorFactory); |
| send_config_.rtp.extensions.push_back(RtpExtension( |
| extension_type.c_str(), kTransmissionTimeOffsetExtensionId)); |
| } |
| |
| test::DirectTransport receiver_transport; |
| StreamObserver stream_observer(rtx_ssrc_map, |
| &receiver_transport, |
| Clock::GetRealTimeClock(), |
| rbe_factory.get(), |
| control_type); |
| |
| Call::Config call_config(&stream_observer); |
| if (start_bitrate_bps != 0) { |
| call_config.start_bitrate_bps = start_bitrate_bps; |
| stream_observer.set_start_bitrate_bps(start_bitrate_bps); |
| } |
| |
| CreateSenderCall(call_config); |
| |
| receiver_transport.SetReceiver(sender_call_->Receiver()); |
| |
| if (num_streams == 1) { |
| video_streams_[0].target_bitrate_bps = 2000000; |
| video_streams_[0].max_bitrate_bps = 2000000; |
| } |
| |
| send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| send_config_.rtp.ssrcs = ssrcs; |
| if (rtx) { |
| send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; |
| send_config_.rtp.rtx.ssrcs = rtx_ssrcs; |
| send_config_.rtp.rtx.pad_with_redundant_payloads = true; |
| } |
| |
| if (num_streams == 1) { |
| // For single stream rampup until 1mbps |
| stream_observer.set_expected_bitrate_bps(kSingleStreamTargetBps); |
| } else { |
| // For multi stream rampup until all streams are being sent. That means |
| // enough birate to send all the target streams plus the min bitrate of |
| // the last one. |
| int expected_bitrate_bps = video_streams_.back().min_bitrate_bps; |
| for (size_t i = 0; i < video_streams_.size() - 1; ++i) { |
| expected_bitrate_bps += video_streams_[i].target_bitrate_bps; |
| } |
| stream_observer.set_expected_bitrate_bps(expected_bitrate_bps); |
| } |
| |
| CreateStreams(); |
| CreateFrameGeneratorCapturer(); |
| |
| Start(); |
| |
| EXPECT_EQ(kEventSignaled, stream_observer.Wait()); |
| |
| Stop(); |
| DestroyStreams(); |
| } |
| |
| void RampUpTest::RunRampUpDownUpTest(size_t number_of_streams, bool rtx) { |
| test::DirectTransport receiver_transport; |
| LowRateStreamObserver stream_observer( |
| &receiver_transport, Clock::GetRealTimeClock(), number_of_streams, rtx); |
| |
| Call::Config call_config(&stream_observer); |
| CreateSenderCall(call_config); |
| receiver_transport.SetReceiver(sender_call_->Receiver()); |
| |
| CreateSendConfig(number_of_streams); |
| |
| send_config_.rtp.nack.rtp_history_ms = kNackRtpHistoryMs; |
| send_config_.rtp.extensions.push_back(RtpExtension( |
| RtpExtension::kTOffset, kTransmissionTimeOffsetExtensionId)); |
| send_config_.suspend_below_min_bitrate = true; |
| if (rtx) { |
| send_config_.rtp.rtx.payload_type = kSendRtxPayloadType; |
| send_config_.rtp.rtx.ssrcs = GenerateSsrcs(number_of_streams, 200); |
| send_config_.rtp.rtx.pad_with_redundant_payloads = true; |
| } |
| |
| CreateStreams(); |
| stream_observer.SetSendStream(send_stream_); |
| |
| CreateFrameGeneratorCapturer(); |
| |
| Start(); |
| |
| EXPECT_EQ(kEventSignaled, stream_observer.Wait()); |
| |
| Stop(); |
| DestroyStreams(); |
| } |
| |
| TEST_F(RampUpTest, SingleStream) { |
| RunRampUpTest(false, 1, 0, RtpExtension::kTOffset); |
| } |
| |
| TEST_F(RampUpTest, Simulcast) { |
| RunRampUpTest(false, 3, 0, RtpExtension::kTOffset); |
| } |
| |
| TEST_F(RampUpTest, SimulcastWithRtx) { |
| RunRampUpTest(true, 3, 0, RtpExtension::kTOffset); |
| } |
| |
| TEST_F(RampUpTest, SingleStreamWithHighStartBitrate) { |
| RunRampUpTest(false, 1, 0.9 * kSingleStreamTargetBps, RtpExtension::kTOffset); |
| } |
| |
| TEST_F(RampUpTest, UpDownUpOneStream) { RunRampUpDownUpTest(1, false); } |
| |
| TEST_F(RampUpTest, UpDownUpThreeStreams) { RunRampUpDownUpTest(3, false); } |
| |
| TEST_F(RampUpTest, UpDownUpOneStreamRtx) { RunRampUpDownUpTest(1, true); } |
| |
| TEST_F(RampUpTest, UpDownUpThreeStreamsRtx) { RunRampUpDownUpTest(3, true); } |
| |
| } // namespace webrtc |
