10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
395 lines
12 KiB
C++
395 lines
12 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 <memory>
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#include <string>
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#include <vector>
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#include "absl/memory/memory.h"
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#include "absl/types/optional.h"
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#include "api/bitrate_constraints.h"
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#include "api/test/simulated_network.h"
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#include "api/test/video_quality_test_fixture.h"
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#include "api/video_codecs/video_codec.h"
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#include "rtc_base/checks.h"
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#include "rtc_base/flags.h"
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#include "rtc_base/logging.h"
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#include "rtc_base/string_encode.h"
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#include "system_wrappers/include/field_trial.h"
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#include "test/field_trial.h"
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#include "test/gtest.h"
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#include "test/run_test.h"
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#include "video/video_quality_test.h"
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namespace webrtc {
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namespace flags {
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// Flags common with video loopback, with different default values.
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WEBRTC_DEFINE_int(width, 1850, "Video width (crops source).");
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size_t Width() {
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return static_cast<size_t>(FLAG_width);
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}
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WEBRTC_DEFINE_int(height, 1110, "Video height (crops source).");
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size_t Height() {
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return static_cast<size_t>(FLAG_height);
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}
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WEBRTC_DEFINE_int(fps, 5, "Frames per second.");
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int Fps() {
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return static_cast<int>(FLAG_fps);
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}
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WEBRTC_DEFINE_int(min_bitrate, 50, "Call and stream min bitrate in kbps.");
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int MinBitrateKbps() {
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return static_cast<int>(FLAG_min_bitrate);
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}
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WEBRTC_DEFINE_int(start_bitrate, 300, "Call start bitrate in kbps.");
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int StartBitrateKbps() {
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return static_cast<int>(FLAG_start_bitrate);
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}
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WEBRTC_DEFINE_int(target_bitrate, 200, "Stream target bitrate in kbps.");
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int TargetBitrateKbps() {
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return static_cast<int>(FLAG_target_bitrate);
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}
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WEBRTC_DEFINE_int(max_bitrate, 1000, "Call and stream max bitrate in kbps.");
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int MaxBitrateKbps() {
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return static_cast<int>(FLAG_max_bitrate);
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}
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WEBRTC_DEFINE_int(num_temporal_layers, 2, "Number of temporal layers to use.");
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int NumTemporalLayers() {
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return static_cast<int>(FLAG_num_temporal_layers);
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}
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// Flags common with video loopback, with equal default values.
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WEBRTC_DEFINE_string(codec, "VP8", "Video codec to use.");
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std::string Codec() {
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return static_cast<std::string>(FLAG_codec);
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}
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WEBRTC_DEFINE_string(rtc_event_log_name,
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"",
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"Filename for rtc event log. Two files "
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"with \"_send\" and \"_recv\" suffixes will be created.");
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std::string RtcEventLogName() {
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return static_cast<std::string>(FLAG_rtc_event_log_name);
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}
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WEBRTC_DEFINE_string(rtp_dump_name,
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"",
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"Filename for dumped received RTP stream.");
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std::string RtpDumpName() {
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return static_cast<std::string>(FLAG_rtp_dump_name);
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}
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WEBRTC_DEFINE_int(
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selected_tl,
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-1,
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"Temporal layer to show or analyze. -1 to disable filtering.");
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int SelectedTL() {
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return static_cast<int>(FLAG_selected_tl);
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}
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WEBRTC_DEFINE_int(
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duration,
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0,
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"Duration of the test in seconds. If 0, rendered will be shown instead.");
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int DurationSecs() {
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return static_cast<int>(FLAG_duration);
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}
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WEBRTC_DEFINE_string(output_filename, "", "Target graph data filename.");
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std::string OutputFilename() {
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return static_cast<std::string>(FLAG_output_filename);
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}
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WEBRTC_DEFINE_string(graph_title,
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"",
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"If empty, title will be generated automatically.");
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std::string GraphTitle() {
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return static_cast<std::string>(FLAG_graph_title);
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}
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WEBRTC_DEFINE_int(loss_percent, 0, "Percentage of packets randomly lost.");
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int LossPercent() {
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return static_cast<int>(FLAG_loss_percent);
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}
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WEBRTC_DEFINE_int(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>(FLAG_link_capacity);
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}
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WEBRTC_DEFINE_int(queue_size,
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0,
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"Size of the bottleneck link queue in packets.");
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int QueueSize() {
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return static_cast<int>(FLAG_queue_size);
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}
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WEBRTC_DEFINE_int(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>(FLAG_avg_propagation_delay_ms);
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}
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WEBRTC_DEFINE_int(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>(FLAG_std_propagation_delay_ms);
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}
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WEBRTC_DEFINE_int(num_streams, 0, "Number of streams to show or analyze.");
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int NumStreams() {
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return static_cast<int>(FLAG_num_streams);
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}
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WEBRTC_DEFINE_int(selected_stream,
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0,
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"ID of the stream to show or analyze. "
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"Set to the number of streams to show them all.");
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int SelectedStream() {
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return static_cast<int>(FLAG_selected_stream);
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}
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WEBRTC_DEFINE_int(num_spatial_layers, 1, "Number of spatial layers to use.");
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int NumSpatialLayers() {
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return static_cast<int>(FLAG_num_spatial_layers);
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}
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WEBRTC_DEFINE_int(
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inter_layer_pred,
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0,
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"Inter-layer prediction mode. "
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"0 - enabled, 1 - disabled, 2 - enabled only for key pictures.");
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InterLayerPredMode InterLayerPred() {
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if (FLAG_inter_layer_pred == 0) {
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return InterLayerPredMode::kOn;
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} else if (FLAG_inter_layer_pred == 1) {
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return InterLayerPredMode::kOff;
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} else {
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RTC_DCHECK_EQ(FLAG_inter_layer_pred, 2);
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return InterLayerPredMode::kOnKeyPic;
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}
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}
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WEBRTC_DEFINE_int(selected_sl,
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-1,
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"Spatial layer to show or analyze. -1 to disable filtering.");
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int SelectedSL() {
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return static_cast<int>(FLAG_selected_sl);
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}
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WEBRTC_DEFINE_string(
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stream0,
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"",
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"Comma separated values describing VideoStream for stream #0.");
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std::string Stream0() {
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return static_cast<std::string>(FLAG_stream0);
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}
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WEBRTC_DEFINE_string(
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stream1,
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"",
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"Comma separated values describing VideoStream for stream #1.");
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std::string Stream1() {
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return static_cast<std::string>(FLAG_stream1);
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}
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WEBRTC_DEFINE_string(
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sl0,
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"",
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"Comma separated values describing SpatialLayer for layer #0.");
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std::string SL0() {
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return static_cast<std::string>(FLAG_sl0);
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}
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WEBRTC_DEFINE_string(
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sl1,
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"",
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"Comma separated values describing SpatialLayer for layer #1.");
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std::string SL1() {
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return static_cast<std::string>(FLAG_sl1);
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}
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WEBRTC_DEFINE_string(
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encoded_frame_path,
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"",
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"The base path for encoded frame logs. Created files will have "
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"the form <encoded_frame_path>.<n>.(recv|send.<m>).ivf");
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std::string EncodedFramePath() {
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return static_cast<std::string>(FLAG_encoded_frame_path);
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}
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WEBRTC_DEFINE_bool(logs, false, "print logs to stderr");
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WEBRTC_DEFINE_bool(send_side_bwe, true, "Use send-side bandwidth estimation");
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WEBRTC_DEFINE_bool(generic_descriptor,
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false,
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"Use the generic frame descriptor.");
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WEBRTC_DEFINE_bool(allow_reordering, false, "Allow packet reordering to occur");
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WEBRTC_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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// Screenshare-specific flags.
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WEBRTC_DEFINE_int(min_transmit_bitrate,
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400,
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"Min transmit bitrate incl. padding.");
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int MinTransmitBitrateKbps() {
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return FLAG_min_transmit_bitrate;
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}
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WEBRTC_DEFINE_bool(
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generate_slides,
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false,
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"Whether to use randomly generated slides or read them from files.");
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bool GenerateSlides() {
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return static_cast<int>(FLAG_generate_slides);
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}
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WEBRTC_DEFINE_int(slide_change_interval,
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10,
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"Interval (in seconds) between simulated slide changes.");
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int SlideChangeInterval() {
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return static_cast<int>(FLAG_slide_change_interval);
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}
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WEBRTC_DEFINE_int(
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scroll_duration,
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0,
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"Duration (in seconds) during which a slide will be scrolled into place.");
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int ScrollDuration() {
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return static_cast<int>(FLAG_scroll_duration);
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}
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WEBRTC_DEFINE_string(
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slides,
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"",
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"Comma-separated list of *.yuv files to display as slides.");
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std::vector<std::string> Slides() {
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std::vector<std::string> slides;
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std::string slides_list = FLAG_slides;
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rtc::tokenize(slides_list, ',', &slides);
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return slides;
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}
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WEBRTC_DEFINE_bool(help, false, "prints this message");
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} // namespace flags
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void Loopback() {
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BuiltInNetworkBehaviorConfig 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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pipe_config.allow_reordering = flags::FLAG_allow_reordering;
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BitrateConstraints 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 = -1; // Don't cap bandwidth estimate.
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VideoQualityTest::Params params;
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params.call = {flags::FLAG_send_side_bwe, flags::FLAG_generic_descriptor,
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call_bitrate_config};
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params.video[0] = {true,
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flags::Width(),
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flags::Height(),
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flags::Fps(),
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flags::MinBitrateKbps() * 1000,
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flags::TargetBitrateKbps() * 1000,
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flags::MaxBitrateKbps() * 1000,
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false,
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flags::Codec(),
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flags::NumTemporalLayers(),
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flags::SelectedTL(),
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flags::MinTransmitBitrateKbps() * 1000,
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false, // ULPFEC disabled.
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false, // FlexFEC disabled.
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false, // Automatic scaling disabled.
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""};
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params.screenshare[0] = {true, flags::GenerateSlides(),
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flags::SlideChangeInterval(),
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flags::ScrollDuration(), flags::Slides()};
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params.analyzer = {"screenshare",
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0.0,
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0.0,
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flags::DurationSecs(),
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flags::OutputFilename(),
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flags::GraphTitle()};
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params.config = pipe_config;
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params.logging = {flags::RtcEventLogName(), flags::RtpDumpName(),
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flags::EncodedFramePath()};
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if (flags::NumStreams() > 1 && flags::Stream0().empty() &&
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flags::Stream1().empty()) {
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params.ss[0].infer_streams = true;
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}
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std::vector<std::string> stream_descriptors;
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stream_descriptors.push_back(flags::Stream0());
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stream_descriptors.push_back(flags::Stream1());
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std::vector<std::string> SL_descriptors;
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SL_descriptors.push_back(flags::SL0());
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SL_descriptors.push_back(flags::SL1());
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VideoQualityTest::FillScalabilitySettings(
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¶ms, 0, stream_descriptors, flags::NumStreams(),
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flags::SelectedStream(), flags::NumSpatialLayers(), flags::SelectedSL(),
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flags::InterLayerPred(), SL_descriptors);
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auto fixture = absl::make_unique<VideoQualityTest>(nullptr);
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if (flags::DurationSecs()) {
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fixture->RunWithAnalyzer(params);
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} else {
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fixture->RunWithRenderers(params);
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}
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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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rtc::FlagList::SetFlagsFromCommandLine(&argc, argv, true);
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if (webrtc::flags::FLAG_help) {
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rtc::FlagList::Print(nullptr, false);
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return 0;
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}
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rtc::LogMessage::SetLogToStderr(webrtc::flags::FLAG_logs);
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webrtc::test::ValidateFieldTrialsStringOrDie(
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webrtc::flags::FLAG_force_fieldtrials);
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// InitFieldTrialsFromString stores the char*, so the char array must outlive
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// the application.
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webrtc::field_trial::InitFieldTrialsFromString(
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webrtc::flags::FLAG_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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