726 lines
26 KiB
C++
726 lines
26 KiB
C++
/*
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* Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#include "call/rampup_tests.h"
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#include <memory>
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#include "absl/flags/flag.h"
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#include "api/rtc_event_log/rtc_event_log_factory.h"
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#include "api/rtc_event_log_output_file.h"
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#include "api/task_queue/default_task_queue_factory.h"
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#include "api/task_queue/task_queue_base.h"
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#include "api/task_queue/task_queue_factory.h"
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#include "call/fake_network_pipe.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/platform_thread.h"
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#include "rtc_base/string_encode.h"
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#include "rtc_base/task_queue_for_test.h"
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#include "rtc_base/time_utils.h"
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#include "test/encoder_settings.h"
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#include "test/field_trial.h"
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#include "test/gtest.h"
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#include "test/testsupport/perf_test.h"
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ABSL_FLAG(std::string,
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ramp_dump_name,
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"",
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"Filename for dumped received RTP stream.");
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namespace webrtc {
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namespace {
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constexpr TimeDelta kPollInterval = TimeDelta::Millis(20);
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static const int kExpectedHighVideoBitrateBps = 80000;
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static const int kExpectedHighAudioBitrateBps = 30000;
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static const int kLowBandwidthLimitBps = 20000;
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// Set target detected bitrate to slightly larger than the target bitrate to
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// avoid flakiness.
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static const int kLowBitrateMarginBps = 2000;
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std::vector<uint32_t> GenerateSsrcs(size_t num_streams, uint32_t ssrc_offset) {
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std::vector<uint32_t> ssrcs;
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for (size_t i = 0; i != num_streams; ++i)
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ssrcs.push_back(static_cast<uint32_t>(ssrc_offset + i));
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return ssrcs;
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}
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} // namespace
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RampUpTester::RampUpTester(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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unsigned int start_bitrate_bps,
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int64_t min_run_time_ms,
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const std::string& extension_type,
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bool rtx,
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bool red,
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bool report_perf_stats,
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TaskQueueBase* task_queue)
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: EndToEndTest(test::CallTest::kLongTimeoutMs),
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clock_(Clock::GetRealTimeClock()),
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num_video_streams_(num_video_streams),
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num_audio_streams_(num_audio_streams),
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num_flexfec_streams_(num_flexfec_streams),
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rtx_(rtx),
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red_(red),
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report_perf_stats_(report_perf_stats),
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sender_call_(nullptr),
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send_stream_(nullptr),
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send_transport_(nullptr),
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send_simulated_network_(nullptr),
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start_bitrate_bps_(start_bitrate_bps),
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min_run_time_ms_(min_run_time_ms),
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expected_bitrate_bps_(0),
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test_start_ms_(-1),
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ramp_up_finished_ms_(-1),
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extension_type_(extension_type),
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video_ssrcs_(GenerateSsrcs(num_video_streams_, 100)),
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video_rtx_ssrcs_(GenerateSsrcs(num_video_streams_, 200)),
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audio_ssrcs_(GenerateSsrcs(num_audio_streams_, 300)),
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task_queue_(task_queue) {
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if (red_)
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EXPECT_EQ(0u, num_flexfec_streams_);
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EXPECT_LE(num_audio_streams_, 1u);
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}
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RampUpTester::~RampUpTester() = default;
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void RampUpTester::ModifySenderBitrateConfig(
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BitrateConstraints* bitrate_config) {
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if (start_bitrate_bps_ != 0) {
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bitrate_config->start_bitrate_bps = start_bitrate_bps_;
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}
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bitrate_config->min_bitrate_bps = 10000;
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}
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void RampUpTester::OnVideoStreamsCreated(
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VideoSendStream* send_stream,
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const std::vector<VideoReceiveStream*>& receive_streams) {
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send_stream_ = send_stream;
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}
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std::unique_ptr<test::PacketTransport> RampUpTester::CreateSendTransport(
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TaskQueueBase* task_queue,
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Call* sender_call) {
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auto network = std::make_unique<SimulatedNetwork>(forward_transport_config_);
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send_simulated_network_ = network.get();
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auto send_transport = std::make_unique<test::PacketTransport>(
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task_queue, sender_call, this, test::PacketTransport::kSender,
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test::CallTest::payload_type_map_,
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std::make_unique<FakeNetworkPipe>(Clock::GetRealTimeClock(),
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std::move(network)));
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send_transport_ = send_transport.get();
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return send_transport;
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}
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size_t RampUpTester::GetNumVideoStreams() const {
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return num_video_streams_;
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}
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size_t RampUpTester::GetNumAudioStreams() const {
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return num_audio_streams_;
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}
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size_t RampUpTester::GetNumFlexfecStreams() const {
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return num_flexfec_streams_;
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}
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class RampUpTester::VideoStreamFactory
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: public VideoEncoderConfig::VideoStreamFactoryInterface {
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public:
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VideoStreamFactory() {}
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private:
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std::vector<VideoStream> CreateEncoderStreams(
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int width,
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int height,
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const VideoEncoderConfig& encoder_config) override {
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std::vector<VideoStream> streams =
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test::CreateVideoStreams(width, height, encoder_config);
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if (encoder_config.number_of_streams == 1) {
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streams[0].target_bitrate_bps = streams[0].max_bitrate_bps = 2000000;
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}
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return streams;
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}
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};
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void RampUpTester::ModifyVideoConfigs(
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VideoSendStream::Config* send_config,
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std::vector<VideoReceiveStream::Config>* receive_configs,
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VideoEncoderConfig* encoder_config) {
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send_config->suspend_below_min_bitrate = true;
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encoder_config->number_of_streams = num_video_streams_;
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encoder_config->max_bitrate_bps = 2000000;
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encoder_config->video_stream_factory =
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new rtc::RefCountedObject<RampUpTester::VideoStreamFactory>();
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if (num_video_streams_ == 1) {
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// For single stream rampup until 1mbps
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expected_bitrate_bps_ = kSingleStreamTargetBps;
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} else {
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// To ensure simulcast rate allocation.
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send_config->rtp.payload_name = "VP8";
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encoder_config->codec_type = kVideoCodecVP8;
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std::vector<VideoStream> streams = test::CreateVideoStreams(
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test::CallTest::kDefaultWidth, test::CallTest::kDefaultHeight,
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*encoder_config);
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// For multi stream rampup until all streams are being sent. That means
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// enough bitrate to send all the target streams plus the min bitrate of
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// the last one.
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expected_bitrate_bps_ = streams.back().min_bitrate_bps;
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for (size_t i = 0; i < streams.size() - 1; ++i) {
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expected_bitrate_bps_ += streams[i].target_bitrate_bps;
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}
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}
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send_config->rtp.extensions.clear();
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bool remb;
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bool transport_cc;
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if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
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remb = true;
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transport_cc = false;
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send_config->rtp.extensions.push_back(
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RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
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} else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
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remb = false;
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transport_cc = true;
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send_config->rtp.extensions.push_back(RtpExtension(
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extension_type_.c_str(), kTransportSequenceNumberExtensionId));
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} else {
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remb = true;
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transport_cc = false;
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send_config->rtp.extensions.push_back(RtpExtension(
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extension_type_.c_str(), kTransmissionTimeOffsetExtensionId));
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}
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send_config->rtp.nack.rtp_history_ms = test::CallTest::kNackRtpHistoryMs;
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send_config->rtp.ssrcs = video_ssrcs_;
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if (rtx_) {
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send_config->rtp.rtx.payload_type = test::CallTest::kSendRtxPayloadType;
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send_config->rtp.rtx.ssrcs = video_rtx_ssrcs_;
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}
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if (red_) {
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send_config->rtp.ulpfec.ulpfec_payload_type =
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test::CallTest::kUlpfecPayloadType;
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send_config->rtp.ulpfec.red_payload_type = test::CallTest::kRedPayloadType;
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if (rtx_) {
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send_config->rtp.ulpfec.red_rtx_payload_type =
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test::CallTest::kRtxRedPayloadType;
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}
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}
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size_t i = 0;
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for (VideoReceiveStream::Config& recv_config : *receive_configs) {
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recv_config.rtp.transport_cc = transport_cc;
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recv_config.rtp.extensions = send_config->rtp.extensions;
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recv_config.decoders.reserve(1);
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recv_config.decoders[0].payload_type = send_config->rtp.payload_type;
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recv_config.decoders[0].video_format =
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SdpVideoFormat(send_config->rtp.payload_name);
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recv_config.rtp.remote_ssrc = video_ssrcs_[i];
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recv_config.rtp.nack.rtp_history_ms = send_config->rtp.nack.rtp_history_ms;
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if (red_) {
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recv_config.rtp.red_payload_type =
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send_config->rtp.ulpfec.red_payload_type;
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recv_config.rtp.ulpfec_payload_type =
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send_config->rtp.ulpfec.ulpfec_payload_type;
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if (rtx_) {
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recv_config.rtp.rtx_associated_payload_types
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[send_config->rtp.ulpfec.red_rtx_payload_type] =
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send_config->rtp.ulpfec.red_payload_type;
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}
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}
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if (rtx_) {
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recv_config.rtp.rtx_ssrc = video_rtx_ssrcs_[i];
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recv_config.rtp
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.rtx_associated_payload_types[send_config->rtp.rtx.payload_type] =
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send_config->rtp.payload_type;
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}
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++i;
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}
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RTC_DCHECK_LE(num_flexfec_streams_, 1);
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if (num_flexfec_streams_ == 1) {
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send_config->rtp.flexfec.payload_type = test::CallTest::kFlexfecPayloadType;
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send_config->rtp.flexfec.ssrc = test::CallTest::kFlexfecSendSsrc;
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send_config->rtp.flexfec.protected_media_ssrcs = {video_ssrcs_[0]};
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}
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}
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void RampUpTester::ModifyAudioConfigs(
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AudioSendStream::Config* send_config,
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std::vector<AudioReceiveStream::Config>* receive_configs) {
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if (num_audio_streams_ == 0)
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return;
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EXPECT_NE(RtpExtension::kTimestampOffsetUri, extension_type_)
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<< "Audio BWE not supported with toffset.";
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EXPECT_NE(RtpExtension::kAbsSendTimeUri, extension_type_)
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<< "Audio BWE not supported with abs-send-time.";
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send_config->rtp.ssrc = audio_ssrcs_[0];
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send_config->rtp.extensions.clear();
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send_config->min_bitrate_bps = 6000;
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send_config->max_bitrate_bps = 60000;
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bool transport_cc = false;
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if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
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transport_cc = true;
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send_config->rtp.extensions.push_back(RtpExtension(
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extension_type_.c_str(), kTransportSequenceNumberExtensionId));
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}
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for (AudioReceiveStream::Config& recv_config : *receive_configs) {
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recv_config.rtp.transport_cc = transport_cc;
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recv_config.rtp.extensions = send_config->rtp.extensions;
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recv_config.rtp.remote_ssrc = send_config->rtp.ssrc;
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}
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}
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void RampUpTester::ModifyFlexfecConfigs(
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std::vector<FlexfecReceiveStream::Config>* receive_configs) {
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if (num_flexfec_streams_ == 0)
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return;
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RTC_DCHECK_EQ(1, num_flexfec_streams_);
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(*receive_configs)[0].payload_type = test::CallTest::kFlexfecPayloadType;
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(*receive_configs)[0].remote_ssrc = test::CallTest::kFlexfecSendSsrc;
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(*receive_configs)[0].protected_media_ssrcs = {video_ssrcs_[0]};
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(*receive_configs)[0].local_ssrc = video_ssrcs_[0];
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if (extension_type_ == RtpExtension::kAbsSendTimeUri) {
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(*receive_configs)[0].transport_cc = false;
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(*receive_configs)[0].rtp_header_extensions.push_back(
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RtpExtension(extension_type_.c_str(), kAbsSendTimeExtensionId));
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} else if (extension_type_ == RtpExtension::kTransportSequenceNumberUri) {
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(*receive_configs)[0].transport_cc = true;
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(*receive_configs)[0].rtp_header_extensions.push_back(RtpExtension(
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extension_type_.c_str(), kTransportSequenceNumberExtensionId));
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}
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}
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void RampUpTester::OnCallsCreated(Call* sender_call, Call* receiver_call) {
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RTC_DCHECK(sender_call);
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sender_call_ = sender_call;
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pending_task_ = RepeatingTaskHandle::Start(task_queue_, [this] {
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PollStats();
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return kPollInterval;
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});
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}
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void RampUpTester::PollStats() {
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RTC_DCHECK_RUN_ON(task_queue_);
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Call::Stats stats = sender_call_->GetStats();
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EXPECT_GE(expected_bitrate_bps_, 0);
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if (stats.send_bandwidth_bps >= expected_bitrate_bps_ &&
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(min_run_time_ms_ == -1 ||
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clock_->TimeInMilliseconds() - test_start_ms_ >= min_run_time_ms_)) {
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ramp_up_finished_ms_ = clock_->TimeInMilliseconds();
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observation_complete_.Set();
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pending_task_.Stop();
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}
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}
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void RampUpTester::ReportResult(const std::string& measurement,
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size_t value,
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const std::string& units) const {
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webrtc::test::PrintResult(
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measurement, "",
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::testing::UnitTest::GetInstance()->current_test_info()->name(), value,
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units, false);
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}
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void RampUpTester::AccumulateStats(const VideoSendStream::StreamStats& stream,
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size_t* total_packets_sent,
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size_t* total_sent,
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size_t* padding_sent,
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size_t* media_sent) const {
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*total_packets_sent += stream.rtp_stats.transmitted.packets +
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stream.rtp_stats.retransmitted.packets +
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stream.rtp_stats.fec.packets;
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*total_sent += stream.rtp_stats.transmitted.TotalBytes() +
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stream.rtp_stats.retransmitted.TotalBytes() +
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stream.rtp_stats.fec.TotalBytes();
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*padding_sent += stream.rtp_stats.transmitted.padding_bytes +
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stream.rtp_stats.retransmitted.padding_bytes +
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stream.rtp_stats.fec.padding_bytes;
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*media_sent += stream.rtp_stats.MediaPayloadBytes();
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}
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void RampUpTester::TriggerTestDone() {
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RTC_DCHECK_GE(test_start_ms_, 0);
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// Stop polling stats.
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// Corner case for field_trials=WebRTC-QuickPerfTest/Enabled/
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SendTask(RTC_FROM_HERE, task_queue_, [this] { pending_task_.Stop(); });
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// TODO(holmer): Add audio send stats here too when those APIs are available.
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if (!send_stream_)
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return;
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VideoSendStream::Stats send_stats = send_stream_->GetStats();
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send_stream_ = nullptr; // To avoid dereferencing a bad pointer.
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size_t total_packets_sent = 0;
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size_t total_sent = 0;
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size_t padding_sent = 0;
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size_t media_sent = 0;
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for (uint32_t ssrc : video_ssrcs_) {
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AccumulateStats(send_stats.substreams[ssrc], &total_packets_sent,
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&total_sent, &padding_sent, &media_sent);
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}
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size_t rtx_total_packets_sent = 0;
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size_t rtx_total_sent = 0;
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size_t rtx_padding_sent = 0;
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size_t rtx_media_sent = 0;
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for (uint32_t rtx_ssrc : video_rtx_ssrcs_) {
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AccumulateStats(send_stats.substreams[rtx_ssrc], &rtx_total_packets_sent,
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&rtx_total_sent, &rtx_padding_sent, &rtx_media_sent);
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}
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if (report_perf_stats_) {
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ReportResult("ramp-up-media-sent", media_sent, "bytes");
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ReportResult("ramp-up-padding-sent", padding_sent, "bytes");
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ReportResult("ramp-up-rtx-media-sent", rtx_media_sent, "bytes");
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ReportResult("ramp-up-rtx-padding-sent", rtx_padding_sent, "bytes");
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if (ramp_up_finished_ms_ >= 0) {
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ReportResult("ramp-up-time", ramp_up_finished_ms_ - test_start_ms_,
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"milliseconds");
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}
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ReportResult("ramp-up-average-network-latency",
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send_transport_->GetAverageDelayMs(), "milliseconds");
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}
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}
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void RampUpTester::PerformTest() {
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test_start_ms_ = clock_->TimeInMilliseconds();
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EXPECT_TRUE(Wait()) << "Timed out while waiting for ramp-up to complete.";
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TriggerTestDone();
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}
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RampUpDownUpTester::RampUpDownUpTester(size_t num_video_streams,
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size_t num_audio_streams,
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size_t num_flexfec_streams,
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unsigned int start_bitrate_bps,
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const std::string& extension_type,
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bool rtx,
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bool red,
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const std::vector<int>& loss_rates,
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bool report_perf_stats,
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TaskQueueBase* task_queue)
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: RampUpTester(num_video_streams,
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num_audio_streams,
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num_flexfec_streams,
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start_bitrate_bps,
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0,
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extension_type,
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rtx,
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red,
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report_perf_stats,
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task_queue),
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link_rates_({4 * GetExpectedHighBitrate() / (3 * 1000),
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kLowBandwidthLimitBps / 1000,
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4 * GetExpectedHighBitrate() / (3 * 1000), 0}),
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test_state_(kFirstRampup),
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next_state_(kTransitionToNextState),
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state_start_ms_(clock_->TimeInMilliseconds()),
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interval_start_ms_(clock_->TimeInMilliseconds()),
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sent_bytes_(0),
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loss_rates_(loss_rates) {
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forward_transport_config_.link_capacity_kbps = link_rates_[test_state_];
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forward_transport_config_.queue_delay_ms = 100;
|
|
forward_transport_config_.loss_percent = loss_rates_[test_state_];
|
|
}
|
|
|
|
RampUpDownUpTester::~RampUpDownUpTester() {}
|
|
|
|
void RampUpDownUpTester::PollStats() {
|
|
if (test_state_ == kTestEnd) {
|
|
pending_task_.Stop();
|
|
}
|
|
|
|
int transmit_bitrate_bps = 0;
|
|
bool suspended = false;
|
|
if (num_video_streams_ > 0 && send_stream_) {
|
|
webrtc::VideoSendStream::Stats stats = send_stream_->GetStats();
|
|
for (const auto& it : stats.substreams) {
|
|
transmit_bitrate_bps += it.second.total_bitrate_bps;
|
|
}
|
|
suspended = stats.suspended;
|
|
}
|
|
if (num_audio_streams_ > 0 && sender_call_) {
|
|
// 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);
|
|
}
|
|
|
|
void RampUpDownUpTester::ModifyReceiverBitrateConfig(
|
|
BitrateConstraints* bitrate_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()
|
|
: task_queue_factory_(CreateDefaultTaskQueueFactory()),
|
|
rtc_event_log_factory_(task_queue_factory_.get()) {
|
|
std::string dump_name(absl::GetFlag(FLAGS_ramp_dump_name));
|
|
if (!dump_name.empty()) {
|
|
send_event_log_ = rtc_event_log_factory_.CreateRtcEventLog(
|
|
RtcEventLog::EncodingType::Legacy);
|
|
recv_event_log_ = rtc_event_log_factory_.CreateRtcEventLog(
|
|
RtcEventLog::EncodingType::Legacy);
|
|
bool event_log_started =
|
|
send_event_log_->StartLogging(
|
|
std::make_unique<RtcEventLogOutputFile>(
|
|
dump_name + ".send.rtc.dat", RtcEventLog::kUnlimitedOutput),
|
|
RtcEventLog::kImmediateOutput) &&
|
|
recv_event_log_->StartLogging(
|
|
std::make_unique<RtcEventLogOutputFile>(
|
|
dump_name + ".recv.rtc.dat", RtcEventLog::kUnlimitedOutput),
|
|
RtcEventLog::kImmediateOutput);
|
|
RTC_DCHECK(event_log_started);
|
|
}
|
|
}
|
|
|
|
private:
|
|
const std::unique_ptr<TaskQueueFactory> task_queue_factory_;
|
|
RtcEventLogFactory rtc_event_log_factory_;
|
|
};
|
|
|
|
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, task_queue());
|
|
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, task_queue());
|
|
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, task_queue());
|
|
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) {
|
|
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
|
|
std::vector<int> loss_rates = {0, 0, 0, 0};
|
|
RampUpDownUpTester test(3, 1, 0, kStartBitrateBps,
|
|
RtpExtension::kTransportSequenceNumberUri, true,
|
|
false, loss_rates, false, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, UpDownUpAudioTransportSequenceNumberRtx) {
|
|
test::ScopedFieldTrials field_trials("WebRTC-Audio-SendSideBwe/Enabled/");
|
|
std::vector<int> loss_rates = {0, 0, 0, 0};
|
|
RampUpDownUpTester test(0, 1, 0, kStartBitrateBps,
|
|
RtpExtension::kTransportSequenceNumberUri, true,
|
|
false, loss_rates, false, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, TOffsetSimulcastRedRtx) {
|
|
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTimestampOffsetUri, true,
|
|
true, true, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, AbsSendTime) {
|
|
RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, false, false,
|
|
false, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, AbsSendTimeSimulcastRedRtx) {
|
|
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kAbsSendTimeUri, true, true,
|
|
true, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, TransportSequenceNumber) {
|
|
RampUpTester test(1, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri,
|
|
false, false, false, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, TransportSequenceNumberSimulcast) {
|
|
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri,
|
|
false, false, false, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, TransportSequenceNumberSimulcastRedRtx) {
|
|
RampUpTester test(3, 0, 0, 0, 0, RtpExtension::kTransportSequenceNumberUri,
|
|
true, true, true, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
|
|
TEST_F(RampUpTest, AudioTransportSequenceNumber) {
|
|
RampUpTester test(0, 1, 0, 300000, 10000,
|
|
RtpExtension::kTransportSequenceNumberUri, false, false,
|
|
false, task_queue());
|
|
RunBaseTest(&test);
|
|
}
|
|
} // namespace webrtc
|