Reason for revert: Temporarily reverting as this causes some issues with perf tests. Especially tests with packet loss no longer works. Original issue's description: > Adding support for simulcast and spatial layers into VideoQualityTest > > The CL includes several changes: > - Adding flags describing the streams and spatial layers. > - Reorganizing the order of the flags, to make them easier to maintain. > - Adding a member .params_ to VideoQualityAnalyzer. > (instead of passing it to every member function manually) > - Updating VideoAnalyzer to support simulcast. > (select appropriate ssrc and fix timestamps which are sometimes increased by 1) > - VP9EncoderImpl already had code for automatic calculation of bitrate for each layer. > Changing to first read bitrates and resolution ratios from the flags, if specified. > If not specified, reverting to the old code are setting the values automatically. > - Changing the parameters in LayerFilteringTransport, replacing > xx_discard_thresholds with selected_xx, to make it easier to use for the end user. > > Committed: https://crrev.com/87f83a9a27d657731ccb54025bc04ccad0da136e > Cr-Commit-Position: refs/heads/master@{#10215} TBR=pbos@webrtc.org,mflodman@webrtc.org,ivica@webrtc.org NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true Review URL: https://codereview.webrtc.org/1397363002 Cr-Commit-Position: refs/heads/master@{#10252}
188 lines
5.6 KiB
C++
188 lines
5.6 KiB
C++
/*
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* Copyright (c) 2015 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 <stdio.h>
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#include "gflags/gflags.h"
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#include "testing/gtest/include/gtest/gtest.h"
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#include "webrtc/test/field_trial.h"
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#include "webrtc/test/run_test.h"
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#include "webrtc/video/video_quality_test.h"
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namespace webrtc {
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namespace flags {
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DEFINE_int32(width, 640, "Video width.");
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size_t Width() {
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return static_cast<size_t>(FLAGS_width);
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}
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DEFINE_int32(height, 480, "Video height.");
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size_t Height() {
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return static_cast<size_t>(FLAGS_height);
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}
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DEFINE_int32(fps, 30, "Frames per second.");
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int Fps() {
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return static_cast<int>(FLAGS_fps);
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}
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DEFINE_int32(min_bitrate, 50, "Call and stream min bitrate in kbps.");
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int MinBitrateKbps() {
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return static_cast<int>(FLAGS_min_bitrate);
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}
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DEFINE_int32(start_bitrate, 300, "Call start bitrate in kbps.");
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int StartBitrateKbps() {
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return static_cast<int>(FLAGS_start_bitrate);
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}
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DEFINE_int32(target_bitrate, 800, "Stream target bitrate in kbps.");
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int TargetBitrateKbps() {
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return static_cast<int>(FLAGS_target_bitrate);
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}
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DEFINE_int32(max_bitrate, 800, "Call and stream max bitrate in kbps.");
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int MaxBitrateKbps() {
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return static_cast<int>(FLAGS_max_bitrate);
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}
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DEFINE_string(codec, "VP8", "Video codec to use.");
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std::string Codec() {
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return static_cast<std::string>(FLAGS_codec);
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}
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DEFINE_int32(loss_percent, 0, "Percentage of packets randomly lost.");
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int LossPercent() {
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return static_cast<int>(FLAGS_loss_percent);
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}
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DEFINE_int32(link_capacity,
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0,
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"Capacity (kbps) of the fake link. 0 means infinite.");
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int LinkCapacityKbps() {
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return static_cast<int>(FLAGS_link_capacity);
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}
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DEFINE_int32(queue_size, 0, "Size of the bottleneck link queue in packets.");
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int QueueSize() {
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return static_cast<int>(FLAGS_queue_size);
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}
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DEFINE_int32(avg_propagation_delay_ms,
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0,
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"Average link propagation delay in ms.");
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int AvgPropagationDelayMs() {
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return static_cast<int>(FLAGS_avg_propagation_delay_ms);
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}
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DEFINE_int32(std_propagation_delay_ms,
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0,
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"Link propagation delay standard deviation in ms.");
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int StdPropagationDelayMs() {
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return static_cast<int>(FLAGS_std_propagation_delay_ms);
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}
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DEFINE_bool(logs, false, "print logs to stderr");
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DEFINE_string(
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force_fieldtrials,
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"",
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"Field trials control experimental feature code which can be forced. "
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"E.g. running with --force_fieldtrials=WebRTC-FooFeature/Enable/"
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" will assign the group Enable to field trial WebRTC-FooFeature. Multiple "
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"trials are separated by \"/\"");
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DEFINE_int32(num_temporal_layers,
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1,
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"Number of temporal layers. Set to 1-4 to override.");
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size_t NumTemporalLayers() {
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return static_cast<size_t>(FLAGS_num_temporal_layers);
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}
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DEFINE_int32(
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tl_discard_threshold,
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0,
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"Discard TLs with id greater or equal the threshold. 0 to disable.");
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size_t TLDiscardThreshold() {
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return static_cast<size_t>(FLAGS_tl_discard_threshold);
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}
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DEFINE_string(clip,
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"",
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"Name of the clip to show. If empty, using chroma generator.");
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std::string Clip() {
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return static_cast<std::string>(FLAGS_clip);
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}
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DEFINE_string(
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output_filename,
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"",
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"Name of a target graph data file. If set, no preview will be shown.");
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std::string OutputFilename() {
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return static_cast<std::string>(FLAGS_output_filename);
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}
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DEFINE_int32(duration, 60, "Duration of the test in seconds.");
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int DurationSecs() {
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return static_cast<int>(FLAGS_duration);
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}
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DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
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} // namespace flags
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void Loopback() {
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FakeNetworkPipe::Config pipe_config;
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pipe_config.loss_percent = flags::LossPercent();
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pipe_config.link_capacity_kbps = flags::LinkCapacityKbps();
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pipe_config.queue_length_packets = flags::QueueSize();
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pipe_config.queue_delay_ms = flags::AvgPropagationDelayMs();
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pipe_config.delay_standard_deviation_ms = flags::StdPropagationDelayMs();
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Call::Config::BitrateConfig call_bitrate_config;
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call_bitrate_config.min_bitrate_bps = flags::MinBitrateKbps() * 1000;
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call_bitrate_config.start_bitrate_bps = flags::StartBitrateKbps() * 1000;
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call_bitrate_config.max_bitrate_bps = flags::MaxBitrateKbps() * 1000;
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std::string clip = flags::Clip();
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std::string graph_title = clip.empty() ? "" : "video " + clip;
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VideoQualityTest::Params params{
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{flags::Width(), flags::Height(), flags::Fps(),
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flags::MinBitrateKbps() * 1000, flags::TargetBitrateKbps() * 1000,
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flags::MaxBitrateKbps() * 1000, flags::Codec(),
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flags::NumTemporalLayers(),
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0, // No min transmit bitrate.
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call_bitrate_config, flags::TLDiscardThreshold(),
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flags::FLAGS_send_side_bwe},
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{clip},
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{}, // Screenshare specific.
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{graph_title, 0.0, 0.0, flags::DurationSecs(), flags::OutputFilename()},
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pipe_config,
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flags::FLAGS_logs};
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VideoQualityTest test;
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if (flags::OutputFilename().empty())
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test.RunWithVideoRenderer(params);
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else
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test.RunWithAnalyzer(params);
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}
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} // namespace webrtc
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int main(int argc, char* argv[]) {
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::testing::InitGoogleTest(&argc, argv);
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google::ParseCommandLineFlags(&argc, &argv, true);
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webrtc::test::InitFieldTrialsFromString(
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webrtc::flags::FLAGS_force_fieldtrials);
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webrtc::test::RunTest(webrtc::Loopback);
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return 0;
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}
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