| /* |
| * 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 <assert.h> |
| |
| #include <map> |
| #include <vector> |
| |
| #include "testing/gtest/include/gtest/gtest.h" |
| |
| #include "webrtc/call.h" |
| #include "webrtc/common.h" |
| #include "webrtc/experiments.h" |
| #include "webrtc/modules/remote_bitrate_estimator/include/remote_bitrate_estimator.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/system_wrappers/interface/event_wrapper.h" |
| #include "webrtc/system_wrappers/interface/scoped_ptr.h" |
| #include "webrtc/test/direct_transport.h" |
| #include "webrtc/test/fake_decoder.h" |
| #include "webrtc/test/fake_encoder.h" |
| #include "webrtc/test/frame_generator_capturer.h" |
| #include "webrtc/test/testsupport/perf_test.h" |
| #include "webrtc/video/transport_adapter.h" |
| |
| namespace webrtc { |
| |
| namespace { |
| static const int kAbsoluteSendTimeExtensionId = 7; |
| static const int kMaxPacketSize = 1500; |
| } |
| |
| class StreamObserver : public newapi::Transport, public RemoteBitrateObserver { |
| public: |
| typedef std::map<uint32_t, int> BytesSentMap; |
| typedef std::map<uint32_t, uint32_t> SsrcMap; |
| StreamObserver(int num_expected_ssrcs, |
| const SsrcMap& rtx_media_ssrcs, |
| newapi::Transport* feedback_transport, |
| Clock* clock) |
| : critical_section_(CriticalSectionWrapper::CreateCriticalSection()), |
| all_ssrcs_sent_(EventWrapper::Create()), |
| rtp_parser_(RtpHeaderParser::Create()), |
| feedback_transport_(feedback_transport), |
| receive_stats_(ReceiveStatistics::Create(clock)), |
| payload_registry_(new RTPPayloadRegistry( |
| -1, RTPPayloadStrategy::CreateStrategy(false))), |
| clock_(clock), |
| num_expected_ssrcs_(num_expected_ssrcs), |
| 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) { |
| // 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(); |
| config.outgoing_transport = &feedback_transport_; |
| rtp_rtcp_.reset(RtpRtcp::CreateRtpRtcp(config)); |
| rtp_rtcp_->SetREMBStatus(true); |
| rtp_rtcp_->SetRTCPStatus(kRtcpNonCompound); |
| rtp_parser_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime, |
| kAbsoluteSendTimeExtensionId); |
| AbsoluteSendTimeRemoteBitrateEstimatorFactory rbe_factory; |
| const uint32_t kRemoteBitrateEstimatorMinBitrateBps = 30000; |
| remote_bitrate_estimator_.reset( |
| rbe_factory.Create(this, clock, kRemoteBitrateEstimatorMinBitrateBps)); |
| } |
| |
| virtual void OnReceiveBitrateChanged(const std::vector<unsigned int>& ssrcs, |
| unsigned int bitrate) { |
| CriticalSectionScoped lock(critical_section_.get()); |
| if (ssrcs.size() == num_expected_ssrcs_ && bitrate >= kExpectedBitrateBps) { |
| if (rtx_media_ssrcs_.empty() || rtx_media_sent_ > 0) { |
| const ::testing::TestInfo* const test_info = |
| ::testing::UnitTest::GetInstance()->current_test_info(); |
| webrtc::test::PrintResult("total-sent", "", test_info->name(), |
| total_sent_, "bytes", false); |
| webrtc::test::PrintResult("padding-sent", "", test_info->name(), |
| padding_sent_, "bytes", false); |
| webrtc::test::PrintResult("rtx-media-sent", "", test_info->name(), |
| rtx_media_sent_, "bytes", false); |
| webrtc::test::PrintResult("total-packets-sent", "", test_info->name(), |
| total_packets_sent_, "packets", false); |
| webrtc::test::PrintResult("padding-packets-sent", "", test_info->name(), |
| padding_packets_sent_, "packets", false); |
| webrtc::test::PrintResult("rtx-packets-sent", "", test_info->name(), |
| rtx_media_packets_sent_, "packets", false); |
| all_ssrcs_sent_->Set(); |
| } |
| } |
| rtp_rtcp_->SetREMBData( |
| bitrate, static_cast<uint8_t>(ssrcs.size()), &ssrcs[0]); |
| rtp_rtcp_->Process(); |
| } |
| |
| virtual bool SendRtp(const uint8_t* packet, size_t length) OVERRIDE { |
| CriticalSectionScoped lock(critical_section_.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; |
| } |
| |
| virtual bool SendRtcp(const uint8_t* packet, size_t length) OVERRIDE { |
| return true; |
| } |
| |
| EventTypeWrapper Wait() { return all_ssrcs_sent_->Wait(120 * 1000); } |
| |
| private: |
| static const unsigned int kExpectedBitrateBps = 1200000; |
| |
| scoped_ptr<CriticalSectionWrapper> critical_section_; |
| scoped_ptr<EventWrapper> all_ssrcs_sent_; |
| scoped_ptr<RtpHeaderParser> rtp_parser_; |
| scoped_ptr<RtpRtcp> rtp_rtcp_; |
| internal::TransportAdapter feedback_transport_; |
| scoped_ptr<ReceiveStatistics> receive_stats_; |
| scoped_ptr<RTPPayloadRegistry> payload_registry_; |
| scoped_ptr<RemoteBitrateEstimator> remote_bitrate_estimator_; |
| Clock* clock_; |
| const size_t num_expected_ssrcs_; |
| SsrcMap rtx_media_ssrcs_; |
| size_t total_sent_; |
| size_t padding_sent_; |
| size_t rtx_media_sent_; |
| int total_packets_sent_; |
| int padding_packets_sent_; |
| int rtx_media_packets_sent_; |
| }; |
| |
| class RampUpTest : public ::testing::TestWithParam<bool> { |
| public: |
| virtual void SetUp() { reserved_ssrcs_.clear(); } |
| |
| protected: |
| void RunRampUpTest(bool pacing, bool rtx) { |
| const size_t kNumberOfStreams = 3; |
| std::vector<uint32_t> ssrcs; |
| for (size_t i = 0; i < kNumberOfStreams; ++i) |
| ssrcs.push_back(static_cast<uint32_t>(i + 1)); |
| uint32_t kRtxSsrcs[kNumberOfStreams] = {111, 112, 113}; |
| StreamObserver::SsrcMap rtx_ssrc_map; |
| if (rtx) { |
| for (size_t i = 0; i < ssrcs.size(); ++i) |
| rtx_ssrc_map[kRtxSsrcs[i]] = ssrcs[i]; |
| } |
| test::DirectTransport receiver_transport; |
| int num_expected_ssrcs = kNumberOfStreams + (rtx ? 1 : 0); |
| StreamObserver stream_observer( |
| num_expected_ssrcs, rtx_ssrc_map, &receiver_transport, |
| Clock::GetRealTimeClock()); |
| |
| Call::Config call_config(&stream_observer); |
| webrtc::Config webrtc_config; |
| call_config.webrtc_config = &webrtc_config; |
| webrtc_config.Set<PaddingStrategy>(new PaddingStrategy(rtx)); |
| scoped_ptr<Call> call(Call::Create(call_config)); |
| VideoSendStream::Config send_config = call->GetDefaultSendConfig(); |
| |
| receiver_transport.SetReceiver(call->Receiver()); |
| |
| test::FakeEncoder encoder(Clock::GetRealTimeClock()); |
| send_config.encoder = &encoder; |
| send_config.internal_source = false; |
| test::FakeEncoder::SetCodecSettings(&send_config.codec, kNumberOfStreams); |
| send_config.codec.plType = 125; |
| send_config.pacing = pacing; |
| send_config.rtp.nack.rtp_history_ms = 1000; |
| send_config.rtp.ssrcs.insert(send_config.rtp.ssrcs.begin(), ssrcs.begin(), |
| ssrcs.end()); |
| if (rtx) { |
| send_config.rtp.rtx.rtx_payload_type = 96; |
| send_config.rtp.rtx.ssrcs.insert(send_config.rtp.rtx.ssrcs.begin(), |
| kRtxSsrcs, |
| kRtxSsrcs + kNumberOfStreams); |
| } |
| send_config.rtp.extensions.push_back( |
| RtpExtension(RtpExtension::kAbsSendTime, kAbsoluteSendTimeExtensionId)); |
| |
| VideoSendStream* send_stream = call->CreateVideoSendStream(send_config); |
| |
| scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer( |
| test::FrameGeneratorCapturer::Create(send_stream->Input(), |
| send_config.codec.width, |
| send_config.codec.height, |
| 30, |
| Clock::GetRealTimeClock())); |
| |
| send_stream->StartSending(); |
| frame_generator_capturer->Start(); |
| |
| EXPECT_EQ(kEventSignaled, stream_observer.Wait()); |
| |
| frame_generator_capturer->Stop(); |
| send_stream->StopSending(); |
| |
| call->DestroyVideoSendStream(send_stream); |
| } |
| std::map<uint32_t, bool> reserved_ssrcs_; |
| }; |
| |
| TEST_F(RampUpTest, WithoutPacing) { |
| RunRampUpTest(false, false); |
| } |
| |
| TEST_F(RampUpTest, WithPacing) { |
| RunRampUpTest(true, false); |
| } |
| |
| TEST_F(RampUpTest, WithPacingAndRtx) { |
| RunRampUpTest(true, true); |
| } |
| |
| } // namespace webrtc |
