| /* |
| * 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 "call/rampup_tests.h" |
| |
| #include "call/fake_network_pipe.h" |
| #include "logging/rtc_event_log/output/rtc_event_log_output_file.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/flags.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/platform_thread.h" |
| #include "rtc_base/stringencode.h" |
| #include "test/encoder_settings.h" |
| #include "test/gtest.h" |
| #include "test/testsupport/perf_test.h" |
| |
| namespace webrtc { |
| namespace { |
| |
| static const int64_t kPollIntervalMs = 20; |
| static const int kExpectedHighVideoBitrateBps = 80000; |
| static const int kExpectedHighAudioBitrateBps = 30000; |
| static const int kLowBandwidthLimitBps = 20000; |
| // Set target detected bitrate to slightly larger than the target bitrate to |
| // avoid flakiness. |
| static const int kLowBitrateMarginBps = 2000; |
| |
| 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 |
| |
| WEBRTC_DEFINE_string(ramp_dump_name, |
| "", |
| "Filename for dumped received RTP stream."); |
| |
| RampUpTester::RampUpTester(size_t num_video_streams, |
| size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| unsigned int start_bitrate_bps, |
| int64_t min_run_time_ms, |
| const std::string& extension_type, |
| bool rtx, |
| bool red, |
| bool report_perf_stats) |
| : EndToEndTest(test::CallTest::kLongTimeoutMs), |
| stop_event_(false, false), |
| clock_(Clock::GetRealTimeClock()), |
| num_video_streams_(num_video_streams), |
| num_audio_streams_(num_audio_streams), |
| num_flexfec_streams_(num_flexfec_streams), |
| rtx_(rtx), |
| red_(red), |
| report_perf_stats_(report_perf_stats), |
| sender_call_(nullptr), |
| send_stream_(nullptr), |
| send_transport_(nullptr), |
| send_simulated_network_(nullptr), |
| start_bitrate_bps_(start_bitrate_bps), |
| min_run_time_ms_(min_run_time_ms), |
| expected_bitrate_bps_(0), |
| test_start_ms_(-1), |
| ramp_up_finished_ms_(-1), |
| extension_type_(extension_type), |
| video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)), |
| video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)), |
| audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)), |
| poller_thread_(&BitrateStatsPollingThread, |
| this, |
| "BitrateStatsPollingThread") { |
| if (red_) |
| EXPECT_EQ(0u, num_flexfec_streams_); |
| EXPECT_LE(num_audio_streams_, 1u); |
| } |
| |
| RampUpTester::~RampUpTester() {} |
| |
| void RampUpTester::ModifySenderCallConfig(Call::Config* config) { |
| if (start_bitrate_bps_ != 0) { |
| config->bitrate_config.start_bitrate_bps = start_bitrate_bps_; |
| } |
| config->bitrate_config.min_bitrate_bps = 10000; |
| } |
| |
| void RampUpTester::OnVideoStreamsCreated( |
| VideoSendStream* send_stream, |
| const std::vector<VideoReceiveStream*>& receive_streams) { |
| send_stream_ = send_stream; |
| } |
| |
| test::PacketTransport* RampUpTester::CreateSendTransport( |
| test::SingleThreadedTaskQueueForTesting* task_queue, |
| Call* sender_call) { |
| auto network = absl::make_unique<SimulatedNetwork>(forward_transport_config_); |
| send_simulated_network_ = network.get(); |
| send_transport_ = 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))); |
| return send_transport_; |
| } |
| |
| size_t RampUpTester::GetNumVideoStreams() const { |
| return num_video_streams_; |
| } |
| |
| size_t RampUpTester::GetNumAudioStreams() const { |
| return num_audio_streams_; |
| } |
| |
| size_t RampUpTester::GetNumFlexfecStreams() const { |
| return num_flexfec_streams_; |
| } |
| |
| class RampUpTester::VideoStreamFactory |
| : public VideoEncoderConfig::VideoStreamFactoryInterface { |
| public: |
| VideoStreamFactory() {} |
| |
| private: |
| std::vector<VideoStream> CreateEncoderStreams( |
| int width, |
| int height, |
| const VideoEncoderConfig& encoder_config) override { |
| std::vector<VideoStream> streams = |
| test::CreateVideoStreams(width, height, encoder_config); |
| if (encoder_config.number_of_streams == 1) { |
| streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000; |
| } |
| return streams; |
| } |
| }; |
| |
| void RampUpTester::ModifyVideoConfigs( |
| VideoSendStream::Config* send_config, |
| std::vector<VideoReceiveStream::Config>* receive_configs, |
| VideoEncoderConfig* encoder_config) { |
| send_config->suspend_below_min_bitrate = true; |
| encoder_config->number_of_streams = num_video_streams_; |
| encoder_config->max_bitrate_bps = 2000000; |
| encoder_config->video_stream_factory = |
| new rtc::RefCountedObject<RampUpTester::VideoStreamFactory>(); |
| if (num_video_streams_ == 1) { |
| // For single stream rampup until 1mbps |
| expected_bitrate_bps_ = kSingleStreamTargetBps; |
| } else { |
| // To ensure simulcast rate allocation. |
| send_config->rtp.payload_name = "VP8"; |
| encoder_config->codec_type = kVideoCodecVP8; |
| std::vector<VideoStream> streams = test::CreateVideoStreams( |
| test::CallTest::kDefaultWidth, test::CallTest::kDefaultHeight, |
| *encoder_config); |
| // For multi stream rampup until all streams are being sent. That means |
| // enough bitrate to send all the target streams plus the min bitrate of |
| // the last one. |
| expected_bitrate_bps_ = streams.back().min_bitrate_bps; |
| for (size_t i = 0; i < streams.size() - 1; ++i) { |
| expected_bitrate_bps_ += streams[i].target_bitrate_bps; |
| } |
| } |
| |
| send_config->rtp.extensions.clear(); |
| |
| bool remb; |
| bool transport_cc; |
| if (extension_type_ == RtpExtension::kAbsSendTimeUri) { |
| remb = true; |
| transport_cc = false; |
| send_config->rtp.extensions.push_back( |
| RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); |
| } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { |
| remb = false; |
| transport_cc = true; |
| send_config->rtp.extensions.push_back(RtpExtension( |
| extension_type_.c_str(), kTransportSequenceNumberExtensionId)); |
| } else { |
| remb = true; |
| transport_cc = false; |
| send_config->rtp.extensions.push_back(RtpExtension( |
| extension_type_.c_str(), kTransmissionTimeOffsetExtensionId)); |
| } |
| |
| send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs; |
| send_config->rtp.ssrcs = video_ssrcs_; |
| if (rtx_) { |
| send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType; |
| send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_; |
| } |
| if (red_) { |
| send_config->rtp.ulpfec.ulpfec_payload_type = |
| test::CallTest::kUlpfecPayloadType; |
| send_config->rtp.ulpfec.red_payload_type = test::CallTest::kRedPayloadType; |
| if (rtx_) { |
| send_config->rtp.ulpfec.red_rtx_payload_type = |
| test::CallTest::kRtxRedPayloadType; |
| } |
| } |
| |
| size_t i = 0; |
| for (VideoReceiveStream::Config& recv_config : *receive_configs) { |
| recv_config.rtp.remb = remb; |
| recv_config.rtp.transport_cc = transport_cc; |
| recv_config.rtp.extensions = send_config->rtp.extensions; |
| recv_config.decoders.reserve(1); |
| recv_config.decoders[0].payload_type = send_config->rtp.payload_type; |
| recv_config.decoders[0].video_format = |
| SdpVideoFormat(send_config->rtp.payload_name); |
| |
| recv_config.rtp.remote_ssrc = video_ssrcs_[i]; |
| recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms; |
| |
| if (red_) { |
| recv_config.rtp.red_payload_type = |
| send_config->rtp.ulpfec.red_payload_type; |
| recv_config.rtp.ulpfec_payload_type = |
| send_config->rtp.ulpfec.ulpfec_payload_type; |
| if (rtx_) { |
| recv_config.rtp.rtx_associated_payload_types |
| [send_config->rtp.ulpfec.red_rtx_payload_type] = |
| send_config->rtp.ulpfec.red_payload_type; |
| } |
| } |
| |
| if (rtx_) { |
| recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i]; |
| recv_config.rtp |
| .rtx_associated_payload_types[send_config->rtp.rtx.payload_type] = |
| send_config->rtp.payload_type; |
| } |
| ++i; |
| } |
| |
| RTC_DCHECK_LE(num_flexfec_streams_, 1); |
| if (num_flexfec_streams_ == 1) { |
| send_config->rtp.flexfec.payload_type = test::CallTest::kFlexfecPayloadType; |
| send_config->rtp.flexfec.ssrc = test::CallTest::kFlexfecSendSsrc; |
| send_config->rtp.flexfec.protected_media_ssrcs = {video_ssrcs_[0]}; |
| } |
| } |
| |
| void RampUpTester::ModifyAudioConfigs( |
| AudioSendStream::Config* send_config, |
| std::vector<AudioReceiveStream::Config>* receive_configs) { |
| if (num_audio_streams_ == 0) |
| return; |
| |
| EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_) |
| << "Audio BWE not supported with toffset."; |
| EXPECT_NE(RtpExtension::kAbsSendTimeUri, extension_type_) |
| << "Audio BWE not supported with abs-send-time."; |
| |
| send_config->rtp.ssrc = audio_ssrcs_[0]; |
| send_config->rtp.extensions.clear(); |
| |
| send_config->min_bitrate_bps = 6000; |
| send_config->max_bitrate_bps = 60000; |
| |
| bool transport_cc = false; |
| if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { |
| transport_cc = true; |
| send_config->rtp.extensions.push_back(RtpExtension( |
| extension_type_.c_str(), kTransportSequenceNumberExtensionId)); |
| } |
| |
| for (AudioReceiveStream::Config& recv_config : *receive_configs) { |
| recv_config.rtp.transport_cc = transport_cc; |
| recv_config.rtp.extensions = send_config->rtp.extensions; |
| recv_config.rtp.remote_ssrc = send_config->rtp.ssrc; |
| } |
| } |
| |
| void RampUpTester::ModifyFlexfecConfigs( |
| std::vector<FlexfecReceiveStream::Config>* receive_configs) { |
| if (num_flexfec_streams_ == 0) |
| return; |
| RTC_DCHECK_EQ(1, num_flexfec_streams_); |
| (*receive_configs)[0].payload_type = test::CallTest::kFlexfecPayloadType; |
| (*receive_configs)[0].remote_ssrc = test::CallTest::kFlexfecSendSsrc; |
| (*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]}; |
| (*receive_configs)[0].local_ssrc = video_ssrcs_[0]; |
| if (extension_type_ == RtpExtension::kAbsSendTimeUri) { |
| (*receive_configs)[0].transport_cc = false; |
| (*receive_configs)[0].rtp_header_extensions.push_back( |
| RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId)); |
| } else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) { |
| (*receive_configs)[0].transport_cc = true; |
| (*receive_configs)[0].rtp_header_extensions.push_back(RtpExtension( |
| extension_type_.c_str(), kTransportSequenceNumberExtensionId)); |
| } |
| } |
| |
| void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) { |
| sender_call_ = sender_call; |
| } |
| |
| void RampUpTester::BitrateStatsPollingThread(void* obj) { |
| static_cast<RampUpTester*>(obj)->PollStats(); |
| } |
| |
| void RampUpTester::PollStats() { |
| do { |
| if (sender_call_) { |
| Call::Stats stats = sender_call_->GetStats(); |
| |
| EXPECT_GE(expected_bitrate_bps_, 0); |
| if (stats.send_bandwidth_bps >= expected_bitrate_bps_ && |
| (min_run_time_ms_ == -1 || |
| clock_->TimeInMilliseconds() - test_start_ms_ >= min_run_time_ms_)) { |
| ramp_up_finished_ms_ = clock_->TimeInMilliseconds(); |
| observation_complete_.Set(); |
| } |
| } |
| } while (!stop_event_.Wait(kPollIntervalMs)); |
| } |
| |
| void RampUpTester::ReportResult(const std::string& measurement, |
| size_t value, |
| const std::string& units) const { |
| webrtc::test::PrintResult( |
| measurement, "", |
| ::testing::UnitTest::GetInstance()->current_test_info()->name(), value, |
| units, false); |
| } |
| |
| void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream, |
| size_t* total_packets_sent, |
| size_t* total_sent, |
| size_t* padding_sent, |
| size_t* media_sent) const { |
| *total_packets_sent += stream.rtp_stats.transmitted.packets + |
| stream.rtp_stats.retransmitted.packets + |
| stream.rtp_stats.fec.packets; |
| *total_sent += stream.rtp_stats.transmitted.TotalBytes() + |
| stream.rtp_stats.retransmitted.TotalBytes() + |
| stream.rtp_stats.fec.TotalBytes(); |
| *padding_sent += stream.rtp_stats.transmitted.padding_bytes + |
| stream.rtp_stats.retransmitted.padding_bytes + |
| stream.rtp_stats.fec.padding_bytes; |
| *media_sent += stream.rtp_stats.MediaPayloadBytes(); |
| } |
| |
| void RampUpTester::TriggerTestDone() { |
| RTC_DCHECK_GE(test_start_ms_, 0); |
| |
| // TODO(holmer): Add audio send stats here too when those APIs are available. |
| if (!send_stream_) |
| return; |
| |
| VideoSendStream::Stats send_stats = send_stream_->GetStats(); |
| |
| size_t total_packets_sent = 0; |
| size_t total_sent = 0; |
| size_t padding_sent = 0; |
| size_t media_sent = 0; |
| for (uint32_t ssrc : video_ssrcs_) { |
| AccumulateStats(send_stats.substreams[ssrc], &total_packets_sent, |
| &total_sent, &padding_sent, &media_sent); |
| } |
| |
| size_t rtx_total_packets_sent = 0; |
| size_t rtx_total_sent = 0; |
| size_t rtx_padding_sent = 0; |
| size_t rtx_media_sent = 0; |
| for (uint32_t rtx_ssrc : video_rtx_ssrcs_) { |
| AccumulateStats(send_stats.substreams[rtx_ssrc], &rtx_total_packets_sent, |
| &rtx_total_sent, &rtx_padding_sent, &rtx_media_sent); |
| } |
| |
| if (report_perf_stats_) { |
| ReportResult("ramp-up-media-sent", media_sent, "bytes"); |
| ReportResult("ramp-up-padding-sent", padding_sent, "bytes"); |
| ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, "bytes"); |
| ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes"); |
| if (ramp_up_finished_ms_ >= 0) { |
| ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_, |
| "milliseconds"); |
| } |
| ReportResult("ramp-up-average-network-latency", |
| send_transport_->GetAverageDelayMs(), "milliseconds"); |
| } |
| } |
| |
| void RampUpTester::PerformTest() { |
| test_start_ms_ = clock_->TimeInMilliseconds(); |
| poller_thread_.Start(); |
| EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete."; |
| TriggerTestDone(); |
| stop_event_.Set(); |
| poller_thread_.Stop(); |
| } |
| |
| RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams, |
| size_t num_audio_streams, |
| size_t num_flexfec_streams, |
| unsigned int start_bitrate_bps, |
| const std::string& extension_type, |
| bool rtx, |
| bool red, |
| const std::vector<int>& loss_rates, |
| bool report_perf_stats) |
| : RampUpTester(num_video_streams, |
| num_audio_streams, |
| num_flexfec_streams, |
| start_bitrate_bps, |
| 0, |
| extension_type, |
| rtx, |
| red, |
| report_perf_stats), |
| link_rates_({4 * GetExpectedHighBitrate() / (3 * 1000), |
| kLowBandwidthLimitBps / 1000, |
| 4 * GetExpectedHighBitrate() / (3 * 1000), 0}), |
| test_state_(kFirstRampup), |
| next_state_(kTransitionToNextState), |
| state_start_ms_(clock_->TimeInMilliseconds()), |
| interval_start_ms_(clock_->TimeInMilliseconds()), |
| sent_bytes_(0), |
| loss_rates_(loss_rates) { |
| forward_transport_config_.link_capacity_kbps = link_rates_[test_state_]; |
| forward_transport_config_.queue_delay_ms = 100; |
| forward_transport_config_.loss_percent = loss_rates_[test_state_]; |
| } |
| |
| RampUpDownUpTester::~RampUpDownUpTester() {} |
| |
| void RampUpDownUpTester::PollStats() { |
| do { |
| int transmit_bitrate_bps = 0; |
| bool suspended = false; |
| if (num_video_streams_ > 0) { |
| webrtc::VideoSendStream::Stats stats = send_stream_->GetStats(); |
| for (auto it : stats.substreams) { |
| transmit_bitrate_bps += it.second.total_bitrate_bps; |
| } |
| suspended = stats.suspended; |
| } |
| if (num_audio_streams_ > 0 && sender_call_ != nullptr) { |
| // An audio send stream doesn't have bitrate stats, so the call send BW is |
| // currently used instead. |
| transmit_bitrate_bps = sender_call_->GetStats().send_bandwidth_bps; |
| } |
| EvolveTestState(transmit_bitrate_bps, suspended); |
| } while (!stop_event_.Wait(kPollIntervalMs)); |
| } |
| |
| void RampUpDownUpTester::ModifyReceiverCallConfig(Call::Config* config) { |
| config->bitrate_config.min_bitrate_bps = 10000; |
| } |
| |
| std::string RampUpDownUpTester::GetModifierString() const { |
| std::string str("_"); |
| if (num_video_streams_ > 0) { |
| str += rtc::ToString(num_video_streams_); |
| str += "stream"; |
| str += (num_video_streams_ > 1 ? "s" : ""); |
| str += "_"; |
| } |
| if (num_audio_streams_ > 0) { |
| str += rtc::ToString(num_audio_streams_); |
| str += "stream"; |
| str += (num_audio_streams_ > 1 ? "s" : ""); |
| str += "_"; |
| } |
| str += (rtx_ ? "" : "no"); |
| str += "rtx_"; |
| str += (red_ ? "" : "no"); |
| str += "red"; |
| return str; |
| } |
| |
| int RampUpDownUpTester::GetExpectedHighBitrate() const { |
| int expected_bitrate_bps = 0; |
| if (num_audio_streams_ > 0) |
| expected_bitrate_bps += kExpectedHighAudioBitrateBps; |
| if (num_video_streams_ > 0) |
| expected_bitrate_bps += kExpectedHighVideoBitrateBps; |
| return expected_bitrate_bps; |
| } |
| |
| size_t RampUpDownUpTester::GetFecBytes() const { |
| size_t flex_fec_bytes = 0; |
| if (num_flexfec_streams_ > 0) { |
| webrtc::VideoSendStream::Stats stats = send_stream_->GetStats(); |
| for (const auto& kv : stats.substreams) |
| flex_fec_bytes += kv.second.rtp_stats.fec.TotalBytes(); |
| } |
| return flex_fec_bytes; |
| } |
| |
| bool RampUpDownUpTester::ExpectingFec() const { |
| return num_flexfec_streams_ > 0 && forward_transport_config_.loss_percent > 0; |
| } |
| |
| void RampUpDownUpTester::EvolveTestState(int bitrate_bps, bool suspended) { |
| int64_t now = clock_->TimeInMilliseconds(); |
| switch (test_state_) { |
| case kFirstRampup: |
| EXPECT_FALSE(suspended); |
| if (bitrate_bps >= GetExpectedHighBitrate()) { |
| if (report_perf_stats_) { |
| webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(), |
| "first_rampup", now - state_start_ms_, "ms", |
| false); |
| } |
| // Apply loss during the transition between states if FEC is enabled. |
| forward_transport_config_.loss_percent = loss_rates_[test_state_]; |
| test_state_ = kTransitionToNextState; |
| next_state_ = kLowRate; |
| } |
| break; |
| case kLowRate: { |
| // Audio streams are never suspended. |
| bool check_suspend_state = num_video_streams_ > 0; |
| if (bitrate_bps < kLowBandwidthLimitBps + kLowBitrateMarginBps && |
| suspended == check_suspend_state) { |
| if (report_perf_stats_) { |
| webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(), |
| "rampdown", now - state_start_ms_, "ms", |
| false); |
| } |
| // Apply loss during the transition between states if FEC is enabled. |
| forward_transport_config_.loss_percent = loss_rates_[test_state_]; |
| test_state_ = kTransitionToNextState; |
| next_state_ = kSecondRampup; |
| } |
| break; |
| } |
| case kSecondRampup: |
| if (bitrate_bps >= GetExpectedHighBitrate() && !suspended) { |
| if (report_perf_stats_) { |
| webrtc::test::PrintResult("ramp_up_down_up", GetModifierString(), |
| "second_rampup", now - state_start_ms_, |
| "ms", false); |
| ReportResult("ramp-up-down-up-average-network-latency", |
| send_transport_->GetAverageDelayMs(), "milliseconds"); |
| } |
| // Apply loss during the transition between states if FEC is enabled. |
| forward_transport_config_.loss_percent = loss_rates_[test_state_]; |
| test_state_ = kTransitionToNextState; |
| next_state_ = kTestEnd; |
| } |
| break; |
| case kTestEnd: |
| observation_complete_.Set(); |
| break; |
| case kTransitionToNextState: |
| if (!ExpectingFec() || GetFecBytes() > 0) { |
| test_state_ = next_state_; |
| forward_transport_config_.link_capacity_kbps = link_rates_[test_state_]; |
| // No loss while ramping up and down as it may affect the BWE |
| // negatively, making the test flaky. |
| forward_transport_config_.loss_percent = 0; |
| state_start_ms_ = now; |
| interval_start_ms_ = now; |
| sent_bytes_ = 0; |
| send_simulated_network_->SetConfig(forward_transport_config_); |
| } |
| break; |
| } |
| } |
| |
| class RampUpTest : public test::CallTest { |
| public: |
| RampUpTest() { |
| std::string dump_name(FLAG_ramp_dump_name); |
| if (!dump_name.empty()) { |
| send_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy); |
| recv_event_log_ = RtcEventLog::Create(RtcEventLog::EncodingType::Legacy); |
| bool event_log_started = |
| send_event_log_->StartLogging( |
| absl::make_unique<RtcEventLogOutputFile>( |
| dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput), |
| RtcEventLog::kImmediateOutput) && |
| recv_event_log_->StartLogging( |
| absl::make_unique<RtcEventLogOutputFile>( |
| dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput), |
| RtcEventLog::kImmediateOutput); |
| RTC_DCHECK(event_log_started); |
| } |
| } |
| }; |
| |
| static const uint32_t kStartBitrateBps = 60000; |
| |
| TEST_F(RampUpTest, UpDownUpAbsSendTimeSimulcastRedRtx) { |
| std::vector<int> loss_rates = {0, 0, 0, 0}; |
| RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, |
| RtpExtension::kAbsSendTimeUri, true, true, loss_rates, |
| true); |
| RunBaseTest(&test); |
| } |
| |
| // TODO(bugs.webrtc.org/8878) |
| #if defined(WEBRTC_MAC) |
| #define MAYBE_UpDownUpTransportSequenceNumberRtx \ |
| DISABLED_UpDownUpTransportSequenceNumberRtx |
| #else |
| #define MAYBE_UpDownUpTransportSequenceNumberRtx \ |
| UpDownUpTransportSequenceNumberRtx |
| #endif |
| TEST_F(RampUpTest, MAYBE_UpDownUpTransportSequenceNumberRtx) { |
| std::vector<int> loss_rates = {0, 0, 0, 0}; |
| RampUpDownUpTester test(3, 0, 0, kStartBitrateBps, |
| RtpExtension::kTransportSequenceNumberUri, true, |
| false, loss_rates, true); |
| RunBaseTest(&test); |
| } |
| |
| // TODO(holmer): Tests which don't report perf stats should be moved to a |
| // different executable since they per definition are not perf tests. |
| // This test is disabled because it crashes on Linux, and is flaky on other |
| // platforms. See: crbug.com/webrtc/7919 |
| TEST_F(RampUpTest, DISABLED_UpDownUpTransportSequenceNumberPacketLoss) { |
| std::vector<int> loss_rates = {20, 0, 0, 0}; |
| RampUpDownUpTester test(1, 0, 1, kStartBitrateBps, |
| RtpExtension::kTransportSequenceNumberUri, true, |
| false, loss_rates, false); |
| RunBaseTest(&test); |
| } |
| |
| // TODO(bugs.webrtc.org/8878) |
| #if defined(WEBRTC_MAC) |
| #define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \ |
| DISABLED_UpDownUpAudioVideoTransportSequenceNumberRtx |
| #else |
| #define MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx \ |
| UpDownUpAudioVideoTransportSequenceNumberRtx |
| #endif |
| TEST_F(RampUpTest, MAYBE_UpDownUpAudioVideoTransportSequenceNumberRtx) { |
| std::vector<int> loss_rates = {0, 0, 0, 0}; |
| RampUpDownUpTester test(3, 1, 0, kStartBitrateBps, |
| RtpExtension::kTransportSequenceNumberUri, true, |
| false, loss_rates, false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) { |
| std::vector<int> loss_rates = {0, 0, 0, 0}; |
| RampUpDownUpTester test(0, 1, 0, kStartBitrateBps, |
| RtpExtension::kTransportSequenceNumberUri, true, |
| false, loss_rates, false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, TOffsetSimulcastRedRtx) { |
| RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTimestampOffsetUri, true, |
| true, true); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, AbsSendTime) { |
| RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, false, false, |
| false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, AbsSendTimeSimulcastRedRtx) { |
| RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, true, true, |
| true); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, TransportSequenceNumber) { |
| RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri, |
| false, false, false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, TransportSequenceNumberSimulcast) { |
| RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri, |
| false, false, false); |
| RunBaseTest(&test); |
| } |
| |
| TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) { |
| RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri, |
| true, true, true); |
| RunBaseTest(&test); |
| } |
| |
| // TODO(bugs.webrtc.org/8878) |
| #if defined(WEBRTC_MAC) |
| #define MAYBE_AudioTransportSequenceNumber DISABLED_AudioTransportSequenceNumber |
| #else |
| #define MAYBE_AudioTransportSequenceNumber AudioTransportSequenceNumber |
| #endif |
| TEST_F(RampUpTest, MAYBE_AudioTransportSequenceNumber) { |
| RampUpTester test(0, 1, 0, 300000, 10000, |
| RtpExtension::kTransportSequenceNumberUri, false, false, |
| false); |
| RunBaseTest(&test); |
| } |
| } // namespace webrtc |
