| /* |
| * 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 "call/fake_network_pipe.h" |
| #include "call/simulated_network.h" |
| #include "test/call_test.h" |
| #include "test/gtest.h" |
| #include "test/rtcp_packet_parser.h" |
| |
| namespace webrtc { |
| |
| class ExtendedReportsEndToEndTest : public test::CallTest {}; |
| |
| class RtcpXrObserver : public test::EndToEndTest { |
| public: |
| RtcpXrObserver(bool enable_rrtr, |
| bool enable_target_bitrate, |
| bool enable_zero_target_bitrate) |
| : EndToEndTest(test::CallTest::kDefaultTimeoutMs), |
| enable_rrtr_(enable_rrtr), |
| enable_target_bitrate_(enable_target_bitrate), |
| enable_zero_target_bitrate_(enable_zero_target_bitrate), |
| sent_rtcp_sr_(0), |
| sent_rtcp_rr_(0), |
| sent_rtcp_rrtr_(0), |
| sent_rtcp_target_bitrate_(false), |
| sent_zero_rtcp_target_bitrate_(false), |
| sent_rtcp_dlrr_(0), |
| send_simulated_network_(nullptr) { |
| forward_transport_config_.link_capacity_kbps = 500; |
| forward_transport_config_.queue_delay_ms = 0; |
| forward_transport_config_.loss_percent = 0; |
| } |
| |
| private: |
| // Receive stream should send RR packets (and RRTR packets if enabled). |
| Action OnReceiveRtcp(const uint8_t* packet, size_t length) override { |
| rtc::CritScope lock(&crit_); |
| test::RtcpPacketParser parser; |
| EXPECT_TRUE(parser.Parse(packet, length)); |
| |
| sent_rtcp_rr_ += parser.receiver_report()->num_packets(); |
| EXPECT_EQ(0, parser.sender_report()->num_packets()); |
| EXPECT_GE(1, parser.xr()->num_packets()); |
| if (parser.xr()->num_packets() > 0) { |
| if (parser.xr()->rrtr()) |
| ++sent_rtcp_rrtr_; |
| EXPECT_FALSE(parser.xr()->dlrr()); |
| } |
| |
| return SEND_PACKET; |
| } |
| // Send stream should send SR packets (and DLRR packets if enabled). |
| Action OnSendRtcp(const uint8_t* packet, size_t length) override { |
| rtc::CritScope lock(&crit_); |
| test::RtcpPacketParser parser; |
| EXPECT_TRUE(parser.Parse(packet, length)); |
| |
| if (parser.sender_ssrc() == test::CallTest::kVideoSendSsrcs[1] && |
| enable_zero_target_bitrate_) { |
| // Reduce bandwidth restriction to disable second stream after it was |
| // enabled for some time. |
| forward_transport_config_.link_capacity_kbps = 200; |
| send_simulated_network_->SetConfig(forward_transport_config_); |
| } |
| |
| sent_rtcp_sr_ += parser.sender_report()->num_packets(); |
| EXPECT_LE(parser.xr()->num_packets(), 1); |
| if (parser.xr()->num_packets() > 0) { |
| EXPECT_FALSE(parser.xr()->rrtr()); |
| if (parser.xr()->dlrr()) |
| ++sent_rtcp_dlrr_; |
| if (parser.xr()->target_bitrate()) { |
| sent_rtcp_target_bitrate_ = true; |
| auto target_bitrates = |
| parser.xr()->target_bitrate()->GetTargetBitrates(); |
| if (target_bitrates.empty()) { |
| sent_zero_rtcp_target_bitrate_ = true; |
| } |
| for (const rtcp::TargetBitrate::BitrateItem& item : target_bitrates) { |
| if (item.target_bitrate_kbps == 0) { |
| sent_zero_rtcp_target_bitrate_ = true; |
| break; |
| } |
| } |
| } |
| } |
| |
| if (sent_rtcp_sr_ > kNumRtcpReportPacketsToObserve && |
| sent_rtcp_rr_ > kNumRtcpReportPacketsToObserve && |
| (sent_rtcp_target_bitrate_ || !enable_target_bitrate_) && |
| (sent_zero_rtcp_target_bitrate_ || !enable_zero_target_bitrate_)) { |
| if (enable_rrtr_) { |
| EXPECT_GT(sent_rtcp_rrtr_, 0); |
| EXPECT_GT(sent_rtcp_dlrr_, 0); |
| } else { |
| EXPECT_EQ(sent_rtcp_rrtr_, 0); |
| EXPECT_EQ(sent_rtcp_dlrr_, 0); |
| } |
| EXPECT_EQ(enable_target_bitrate_, sent_rtcp_target_bitrate_); |
| EXPECT_EQ(enable_zero_target_bitrate_, sent_zero_rtcp_target_bitrate_); |
| observation_complete_.Set(); |
| } |
| return SEND_PACKET; |
| } |
| |
| size_t GetNumVideoStreams() const override { |
| // When sending a zero target bitrate, we use two spatial layers so that |
| // we'll still have a layer with non-zero bitrate. |
| return enable_zero_target_bitrate_ ? 2 : 1; |
| } |
| |
| test::PacketTransport* CreateSendTransport( |
| test::SingleThreadedTaskQueueForTesting* task_queue, |
| Call* sender_call) { |
| auto network = |
| absl::make_unique<SimulatedNetwork>(forward_transport_config_); |
| send_simulated_network_ = network.get(); |
| return new test::PacketTransport( |
| task_queue, sender_call, this, test::PacketTransport::kSender, |
| test::CallTest::payload_type_map_, |
| absl::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(), |
| std::move(network))); |
| } |
| |
| void ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) override { |
| if (enable_zero_target_bitrate_) { |
| // Configure VP8 to be able to use simulcast. |
| send_config->rtp.payload_name = "VP8"; |
| encoder_config->codec_type = kVideoCodecVP8; |
| (*receive_configs)[0].decoders.resize(1); |
| (*receive_configs)[0].decoders[0].payload_type = |
| send_config->rtp.payload_type; |
| (*receive_configs)[0].decoders[0].video_format = |
| SdpVideoFormat(send_config->rtp.payload_name); |
| } |
| if (enable_target_bitrate_) { |
| // TargetBitrate only signaled for screensharing. |
| encoder_config->content_type = VideoEncoderConfig::ContentType::kScreen; |
| } |
| (*receive_configs)[0].rtp.rtcp_mode = RtcpMode::kReducedSize; |
| (*receive_configs)[0].rtp.rtcp_xr.receiver_reference_time_report = |
| enable_rrtr_; |
| } |
| |
| void PerformTest() override { |
| EXPECT_TRUE(Wait()) |
| << "Timed out while waiting for RTCP SR/RR packets to be sent."; |
| } |
| |
| static const int kNumRtcpReportPacketsToObserve = 5; |
| |
| rtc::CriticalSection crit_; |
| const bool enable_rrtr_; |
| const bool enable_target_bitrate_; |
| const bool enable_zero_target_bitrate_; |
| int sent_rtcp_sr_; |
| int sent_rtcp_rr_ RTC_GUARDED_BY(&crit_); |
| int sent_rtcp_rrtr_ RTC_GUARDED_BY(&crit_); |
| bool sent_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_); |
| bool sent_zero_rtcp_target_bitrate_ RTC_GUARDED_BY(&crit_); |
| int sent_rtcp_dlrr_; |
| BuiltInNetworkBehaviorConfig forward_transport_config_; |
| SimulatedNetwork* send_simulated_network_; |
| }; |
| |
| TEST_F(ExtendedReportsEndToEndTest, |
| TestExtendedReportsWithRrtrWithoutTargetBitrate) { |
| RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/false, |
| /*enable_zero_target_bitrate=*/false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(ExtendedReportsEndToEndTest, |
| TestExtendedReportsWithoutRrtrWithoutTargetBitrate) { |
| RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/false, |
| /*enable_zero_target_bitrate=*/false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(ExtendedReportsEndToEndTest, |
| TestExtendedReportsWithRrtrWithTargetBitrate) { |
| RtcpXrObserver test(/*enable_rrtr=*/true, /*enable_target_bitrate=*/true, |
| /*enable_zero_target_bitrate=*/false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(ExtendedReportsEndToEndTest, |
| TestExtendedReportsWithoutRrtrWithTargetBitrate) { |
| RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true, |
| /*enable_zero_target_bitrate=*/false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(ExtendedReportsEndToEndTest, |
| TestExtendedReportsCanSignalZeroTargetBitrate) { |
| RtcpXrObserver test(/*enable_rrtr=*/false, /*enable_target_bitrate=*/true, |
| /*enable_zero_target_bitrate=*/true); |
| RunBaseTest(&test); |
| } |
| } // namespace webrtc |
