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/*
* 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 "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 screenshare loopback, with different default values.
WEBRTC_DEFINE_int(width, 640, "Video width.");
size_t Width() {
return static_cast<size_t>(FLAG_width);
}
WEBRTC_DEFINE_int(height, 480, "Video height.");
size_t Height() {
return static_cast<size_t>(FLAG_height);
}
WEBRTC_DEFINE_int(fps, 30, "Frames per second.");
int Fps() {
return static_cast<int>(FLAG_fps);
}
WEBRTC_DEFINE_int(capture_device_index, 0, "Capture device to select");
size_t GetCaptureDevice() {
return static_cast<size_t>(FLAG_capture_device_index);
}
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, 800, "Stream target bitrate in kbps.");
int TargetBitrateKbps() {
return static_cast<int>(FLAG_target_bitrate);
}
WEBRTC_DEFINE_int(max_bitrate, 800, "Call and stream max bitrate in kbps.");
int MaxBitrateKbps() {
return static_cast<int>(FLAG_max_bitrate);
}
WEBRTC_DEFINE_bool(suspend_below_min_bitrate,
false,
"Suspends video below the configured min bitrate.");
WEBRTC_DEFINE_int(num_temporal_layers,
1,
"Number of temporal layers. Set to 1-4 to override.");
int NumTemporalLayers() {
return static_cast<int>(FLAG_num_temporal_layers);
}
WEBRTC_DEFINE_int(
inter_layer_pred,
2,
"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;
}
}
// Flags common with screenshare 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_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(avg_burst_loss_length,
-1,
"Average burst length of lost packets.");
int AvgBurstLossLength() {
return static_cast<int>(FLAG_avg_burst_loss_length);
}
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_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(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(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_bool(use_ulpfec,
false,
"Use RED+ULPFEC forward error correction.");
WEBRTC_DEFINE_bool(use_flexfec, false, "Use FlexFEC forward error correction.");
WEBRTC_DEFINE_bool(audio, false, "Add audio stream");
WEBRTC_DEFINE_bool(
use_real_adm,
false,
"Use real ADM instead of fake (no effect if audio is false)");
WEBRTC_DEFINE_bool(audio_video_sync,
false,
"Sync audio and video stream (no effect if"
" audio is false)");
WEBRTC_DEFINE_bool(audio_dtx,
false,
"Enable audio DTX (no effect if audio is false)");
WEBRTC_DEFINE_bool(video, true, "Add video stream");
WEBRTC_DEFINE_string(
force_fieldtrials,
"",
"Field trials control experimental feature code which can be forced. "
"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enabled/"
" will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
"trials are separated by \"/\"");
// Video-specific flags.
WEBRTC_DEFINE_string(
clip,
"",
"Name of the clip to show. If empty, using chroma generator.");
std::string Clip() {
return static_cast<std::string>(FLAG_clip);
}
WEBRTC_DEFINE_bool(help, false, "prints this message");
} // namespace flags
void Loopback() {
BuiltInNetworkBehaviorConfig pipe_config;
pipe_config.loss_percent = flags::LossPercent();
pipe_config.avg_burst_loss_length = flags::AvgBurstLossLength();
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, 0};
params.video[0] = {flags::FLAG_video,
flags::Width(),
flags::Height(),
flags::Fps(),
flags::MinBitrateKbps() * 1000,
flags::TargetBitrateKbps() * 1000,
flags::MaxBitrateKbps() * 1000,
flags::FLAG_suspend_below_min_bitrate,
flags::Codec(),
flags::NumTemporalLayers(),
flags::SelectedTL(),
0, // No min transmit bitrate.
flags::FLAG_use_ulpfec,
flags::FLAG_use_flexfec,
flags::NumStreams() < 2, // Automatic quality scaling.
flags::Clip(),
flags::GetCaptureDevice()};
params.audio = {flags::FLAG_audio, flags::FLAG_audio_video_sync,
flags::FLAG_audio_dtx, flags::FLAG_use_real_adm};
params.logging = {flags::FLAG_rtc_event_log_name, flags::FLAG_rtp_dump_name,
flags::FLAG_encoded_frame_path};
params.screenshare[0].enabled = false;
params.analyzer = {"video",
0.0,
0.0,
flags::DurationSecs(),
flags::OutputFilename(),
flags::GraphTitle()};
params.config = pipe_config;
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(
&params, 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;
}