| /* |
| * Copyright (c) 2015 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 <stdio.h> |
| #include <memory> |
| #include <string> |
| #include <vector> |
| |
| #include "absl/memory/memory.h" |
| #include "absl/types/optional.h" |
| #include "api/bitrate_constraints.h" |
| #include "api/test/simulated_network.h" |
| #include "api/test/video_quality_test_fixture.h" |
| #include "api/video_codecs/video_codec.h" |
| #include "rtc_base/checks.h" |
| #include "rtc_base/flags.h" |
| #include "rtc_base/logging.h" |
| #include "rtc_base/string_encode.h" |
| #include "system_wrappers/include/field_trial.h" |
| #include "test/field_trial.h" |
| #include "test/gtest.h" |
| #include "test/run_test.h" |
| #include "video/video_quality_test.h" |
| |
| namespace webrtc { |
| namespace flags { |
| |
| // Flags common with video loopback, with different default values. |
| WEBRTC_DEFINE_int(width, 1850, "Video width (crops source)."); |
| size_t Width() { |
| return static_cast<size_t>(FLAG_width); |
| } |
| |
| WEBRTC_DEFINE_int(height, 1110, "Video height (crops source)."); |
| size_t Height() { |
| return static_cast<size_t>(FLAG_height); |
| } |
| |
| WEBRTC_DEFINE_int(fps, 5, "Frames per second."); |
| int Fps() { |
| return static_cast<int>(FLAG_fps); |
| } |
| |
| WEBRTC_DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps."); |
| int MinBitrateKbps() { |
| return static_cast<int>(FLAG_min_bitrate); |
| } |
| |
| WEBRTC_DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps."); |
| int StartBitrateKbps() { |
| return static_cast<int>(FLAG_start_bitrate); |
| } |
| |
| WEBRTC_DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps."); |
| int TargetBitrateKbps() { |
| return static_cast<int>(FLAG_target_bitrate); |
| } |
| |
| WEBRTC_DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps."); |
| int MaxBitrateKbps() { |
| return static_cast<int>(FLAG_max_bitrate); |
| } |
| |
| WEBRTC_DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use."); |
| int NumTemporalLayers() { |
| return static_cast<int>(FLAG_num_temporal_layers); |
| } |
| |
| // Flags common with video loopback, with equal default values. |
| WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use."); |
| std::string Codec() { |
| return static_cast<std::string>(FLAG_codec); |
| } |
| |
| WEBRTC_DEFINE_string(rtc_event_log_name, |
| "", |
| "Filename for rtc event log. Two files " |
| "with \"_send\" and \"_recv\" suffixes will be created."); |
| std::string RtcEventLogName() { |
| return static_cast<std::string>(FLAG_rtc_event_log_name); |
| } |
| |
| WEBRTC_DEFINE_string(rtp_dump_name, |
| "", |
| "Filename for dumped received RTP stream."); |
| std::string RtpDumpName() { |
| return static_cast<std::string>(FLAG_rtp_dump_name); |
| } |
| |
| WEBRTC_DEFINE_int( |
| selected_tl, |
| -1, |
| "Temporal layer to show or analyze. -1 to disable filtering."); |
| int SelectedTL() { |
| return static_cast<int>(FLAG_selected_tl); |
| } |
| |
| WEBRTC_DEFINE_int( |
| duration, |
| 0, |
| "Duration of the test in seconds. If 0, rendered will be shown instead."); |
| int DurationSecs() { |
| return static_cast<int>(FLAG_duration); |
| } |
| |
| WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename."); |
| std::string OutputFilename() { |
| return static_cast<std::string>(FLAG_output_filename); |
| } |
| |
| WEBRTC_DEFINE_string(graph_title, |
| "", |
| "If empty, title will be generated automatically."); |
| std::string GraphTitle() { |
| return static_cast<std::string>(FLAG_graph_title); |
| } |
| |
| WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost."); |
| int LossPercent() { |
| return static_cast<int>(FLAG_loss_percent); |
| } |
| |
| WEBRTC_DEFINE_int(link_capacity, |
| 0, |
| "Capacity (kbps) of the fake link. 0 means infinite."); |
| int LinkCapacityKbps() { |
| return static_cast<int>(FLAG_link_capacity); |
| } |
| |
| WEBRTC_DEFINE_int(queue_size, |
| 0, |
| "Size of the bottleneck link queue in packets."); |
| int QueueSize() { |
| return static_cast<int>(FLAG_queue_size); |
| } |
| |
| WEBRTC_DEFINE_int(avg_propagation_delay_ms, |
| 0, |
| "Average link propagation delay in ms."); |
| int AvgPropagationDelayMs() { |
| return static_cast<int>(FLAG_avg_propagation_delay_ms); |
| } |
| |
| WEBRTC_DEFINE_int(std_propagation_delay_ms, |
| 0, |
| "Link propagation delay standard deviation in ms."); |
| int StdPropagationDelayMs() { |
| return static_cast<int>(FLAG_std_propagation_delay_ms); |
| } |
| |
| WEBRTC_DEFINE_int(num_streams, 0, "Number of streams to show or analyze."); |
| int NumStreams() { |
| return static_cast<int>(FLAG_num_streams); |
| } |
| |
| WEBRTC_DEFINE_int(selected_stream, |
| 0, |
| "ID of the stream to show or analyze. " |
| "Set to the number of streams to show them all."); |
| int SelectedStream() { |
| return static_cast<int>(FLAG_selected_stream); |
| } |
| |
| WEBRTC_DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use."); |
| int NumSpatialLayers() { |
| return static_cast<int>(FLAG_num_spatial_layers); |
| } |
| |
| WEBRTC_DEFINE_int( |
| inter_layer_pred, |
| 0, |
| "Inter-layer prediction mode. " |
| "0 - enabled, 1 - disabled, 2 - enabled only for key pictures."); |
| InterLayerPredMode InterLayerPred() { |
| if (FLAG_inter_layer_pred == 0) { |
| return InterLayerPredMode::kOn; |
| } else if (FLAG_inter_layer_pred == 1) { |
| return InterLayerPredMode::kOff; |
| } else { |
| RTC_DCHECK_EQ(FLAG_inter_layer_pred, 2); |
| return InterLayerPredMode::kOnKeyPic; |
| } |
| } |
| |
| WEBRTC_DEFINE_int(selected_sl, |
| -1, |
| "Spatial layer to show or analyze. -1 to disable filtering."); |
| int SelectedSL() { |
| return static_cast<int>(FLAG_selected_sl); |
| } |
| |
| WEBRTC_DEFINE_string( |
| stream0, |
| "", |
| "Comma separated values describing VideoStream for stream #0."); |
| std::string Stream0() { |
| return static_cast<std::string>(FLAG_stream0); |
| } |
| |
| WEBRTC_DEFINE_string( |
| stream1, |
| "", |
| "Comma separated values describing VideoStream for stream #1."); |
| std::string Stream1() { |
| return static_cast<std::string>(FLAG_stream1); |
| } |
| |
| WEBRTC_DEFINE_string( |
| sl0, |
| "", |
| "Comma separated values describing SpatialLayer for layer #0."); |
| std::string SL0() { |
| return static_cast<std::string>(FLAG_sl0); |
| } |
| |
| WEBRTC_DEFINE_string( |
| sl1, |
| "", |
| "Comma separated values describing SpatialLayer for layer #1."); |
| std::string SL1() { |
| return static_cast<std::string>(FLAG_sl1); |
| } |
| |
| WEBRTC_DEFINE_string( |
| encoded_frame_path, |
| "", |
| "The base path for encoded frame logs. Created files will have " |
| "the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf"); |
| std::string EncodedFramePath() { |
| return static_cast<std::string>(FLAG_encoded_frame_path); |
| } |
| |
| WEBRTC_DEFINE_bool(logs, false, "print logs to stderr"); |
| |
| WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation"); |
| |
| WEBRTC_DEFINE_bool(generic_descriptor, |
| false, |
| "Use the generic frame descriptor."); |
| |
| WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur"); |
| |
| WEBRTC_DEFINE_string( |
| force_fieldtrials, |
| "", |
| "Field trials control experimental feature code which can be forced. " |
| "E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/" |
| " will assign the group Enable to field trial WebRTC-FooFeature. Multiple " |
| "trials are separated by \"/\""); |
| |
| // Screenshare-specific flags. |
| WEBRTC_DEFINE_int(min_transmit_bitrate, |
| 400, |
| "Min transmit bitrate incl. padding."); |
| int MinTransmitBitrateKbps() { |
| return FLAG_min_transmit_bitrate; |
| } |
| |
| WEBRTC_DEFINE_bool( |
| generate_slides, |
| false, |
| "Whether to use randomly generated slides or read them from files."); |
| bool GenerateSlides() { |
| return static_cast<int>(FLAG_generate_slides); |
| } |
| |
| WEBRTC_DEFINE_int(slide_change_interval, |
| 10, |
| "Interval (in seconds) between simulated slide changes."); |
| int SlideChangeInterval() { |
| return static_cast<int>(FLAG_slide_change_interval); |
| } |
| |
| WEBRTC_DEFINE_int( |
| scroll_duration, |
| 0, |
| "Duration (in seconds) during which a slide will be scrolled into place."); |
| int ScrollDuration() { |
| return static_cast<int>(FLAG_scroll_duration); |
| } |
| |
| WEBRTC_DEFINE_string( |
| slides, |
| "", |
| "Comma-separated list of *.yuv files to display as slides."); |
| std::vector<std::string> Slides() { |
| std::vector<std::string> slides; |
| std::string slides_list = FLAG_slides; |
| rtc::tokenize(slides_list, ',', &slides); |
| return slides; |
| } |
| |
| WEBRTC_DEFINE_bool(help, false, "prints this message"); |
| |
| WEBRTC_DEFINE_bool(partial_updates, |
| false, |
| "Pass only changed regions from the " |
| "capturer"); |
| |
| } // namespace flags |
| |
| void Loopback() { |
| BuiltInNetworkBehaviorConfig pipe_config; |
| pipe_config.loss_percent = flags::LossPercent(); |
| pipe_config.link_capacity_kbps = flags::LinkCapacityKbps(); |
| pipe_config.queue_length_packets = flags::QueueSize(); |
| pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs(); |
| pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs(); |
| pipe_config.allow_reordering = flags::FLAG_allow_reordering; |
| |
| BitrateConstraints call_bitrate_config; |
| call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000; |
| call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000; |
| call_bitrate_config.max_bitrate_bps = -1; // Don't cap bandwidth estimate. |
| |
| VideoQualityTest::Params params; |
| params.call = {flags::FLAG_send_side_bwe, flags::FLAG_generic_descriptor, |
| call_bitrate_config}; |
| params.video[0] = {true, |
| flags::Width(), |
| flags::Height(), |
| flags::Fps(), |
| flags::MinBitrateKbps() * 1000, |
| flags::TargetBitrateKbps() * 1000, |
| flags::MaxBitrateKbps() * 1000, |
| false, |
| flags::Codec(), |
| flags::NumTemporalLayers(), |
| flags::SelectedTL(), |
| flags::MinTransmitBitrateKbps() * 1000, |
| false, // ULPFEC disabled. |
| false, // FlexFEC disabled. |
| false, // Automatic scaling disabled. |
| "", |
| 0, // capture_device_index. |
| SdpVideoFormat::Parameters(), |
| flags::FLAG_partial_updates}; |
| params.screenshare[0] = {true, flags::GenerateSlides(), |
| flags::SlideChangeInterval(), |
| flags::ScrollDuration(), flags::Slides()}; |
| params.analyzer = {"screenshare", |
| 0.0, |
| 0.0, |
| flags::DurationSecs(), |
| flags::OutputFilename(), |
| flags::GraphTitle()}; |
| params.config = pipe_config; |
| params.logging = {flags::RtcEventLogName(), flags::RtpDumpName(), |
| flags::EncodedFramePath()}; |
| |
| if (flags::NumStreams() > 1 && flags::Stream0().empty() && |
| flags::Stream1().empty()) { |
| params.ss[0].infer_streams = true; |
| } |
| |
| std::vector<std::string> stream_descriptors; |
| stream_descriptors.push_back(flags::Stream0()); |
| stream_descriptors.push_back(flags::Stream1()); |
| std::vector<std::string> SL_descriptors; |
| SL_descriptors.push_back(flags::SL0()); |
| SL_descriptors.push_back(flags::SL1()); |
| VideoQualityTest::FillScalabilitySettings( |
| ¶ms, 0, stream_descriptors, flags::NumStreams(), |
| flags::SelectedStream(), flags::NumSpatialLayers(), flags::SelectedSL(), |
| flags::InterLayerPred(), SL_descriptors); |
| |
| auto fixture = absl::make_unique<VideoQualityTest>(nullptr); |
| if (flags::DurationSecs()) { |
| fixture->RunWithAnalyzer(params); |
| } else { |
| fixture->RunWithRenderers(params); |
| } |
| } |
| } // namespace webrtc |
| |
| int main(int argc, char* argv[]) { |
| ::testing::InitGoogleTest(&argc, argv); |
| rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true); |
| if (webrtc::flags::FLAG_help) { |
| rtc::FlagList::Print(nullptr, false); |
| return 0; |
| } |
| |
| rtc::LogMessage::SetLogToStderr(webrtc::flags::FLAG_logs); |
| |
| webrtc::test::ValidateFieldTrialsStringOrDie( |
| webrtc::flags::FLAG_force_fieldtrials); |
| // InitFieldTrialsFromString stores the char*, so the char array must outlive |
| // the application. |
| webrtc::field_trial::InitFieldTrialsFromString( |
| webrtc::flags::FLAG_force_fieldtrials); |
| |
| webrtc::test::RunTest(webrtc::Loopback); |
| return 0; |
| } |