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268 lines
10 KiB
268 lines
10 KiB
/*
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* Copyright 2019 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 <atomic>
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#include "test/field_trial.h"
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#include "test/gtest.h"
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#include "test/scenario/scenario.h"
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namespace webrtc {
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namespace test {
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namespace {
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using Capture = VideoStreamConfig::Source::Capture;
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using ContentType = VideoStreamConfig::Encoder::ContentType;
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using Codec = VideoStreamConfig::Encoder::Codec;
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using CodecImpl = VideoStreamConfig::Encoder::Implementation;
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} // namespace
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TEST(VideoStreamTest, ReceivesFramesFromFileBasedStreams) {
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TimeDelta kRunTime = TimeDelta::Millis(500);
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std::vector<int> kFrameRates = {15, 30};
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std::deque<std::atomic<int>> frame_counts(2);
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frame_counts[0] = 0;
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frame_counts[1] = 0;
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{
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Scenario s;
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auto route =
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s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())},
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s.CreateClient("callee", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())});
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s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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c->hooks.frame_pair_handlers = {
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[&](const VideoFramePair&) { frame_counts[0]++; }};
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c->source.capture = Capture::kVideoFile;
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c->source.video_file.name = "foreman_cif";
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c->source.video_file.width = 352;
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c->source.video_file.height = 288;
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c->source.framerate = kFrameRates[0];
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c->encoder.implementation = CodecImpl::kSoftware;
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c->encoder.codec = Codec::kVideoCodecVP8;
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});
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s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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c->hooks.frame_pair_handlers = {
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[&](const VideoFramePair&) { frame_counts[1]++; }};
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c->source.capture = Capture::kImageSlides;
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c->source.slides.images.crop.width = 320;
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c->source.slides.images.crop.height = 240;
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c->source.framerate = kFrameRates[1];
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c->encoder.implementation = CodecImpl::kSoftware;
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c->encoder.codec = Codec::kVideoCodecVP9;
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});
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s.RunFor(kRunTime);
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}
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std::vector<int> expected_counts;
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for (int fps : kFrameRates)
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expected_counts.push_back(
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static_cast<int>(kRunTime.seconds<double>() * fps * 0.8));
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EXPECT_GE(frame_counts[0], expected_counts[0]);
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EXPECT_GE(frame_counts[1], expected_counts[1]);
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}
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TEST(VideoStreamTest, RecievesVp8SimulcastFrames) {
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TimeDelta kRunTime = TimeDelta::Millis(500);
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int kFrameRate = 30;
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std::deque<std::atomic<int>> frame_counts(3);
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frame_counts[0] = 0;
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frame_counts[1] = 0;
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frame_counts[2] = 0;
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{
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Scenario s;
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auto route =
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s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())},
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s.CreateClient("callee", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())});
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s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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// TODO(srte): Replace with code checking for all simulcast streams when
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// there's a hook available for that.
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c->hooks.frame_pair_handlers = {[&](const VideoFramePair& info) {
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frame_counts[info.layer_id]++;
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RTC_DCHECK(info.decoded);
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printf("%i: [%3i->%3i, %i], %i->%i, \n", info.layer_id, info.capture_id,
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info.decode_id, info.repeated, info.captured->width(),
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info.decoded->width());
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}};
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c->source.framerate = kFrameRate;
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// The resolution must be high enough to allow smaller layers to be
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// created.
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c->source.generator.width = 1024;
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c->source.generator.height = 768;
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c->encoder.implementation = CodecImpl::kSoftware;
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c->encoder.codec = Codec::kVideoCodecVP8;
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// By enabling multiple spatial layers, simulcast will be enabled for VP8.
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c->encoder.layers.spatial = 3;
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});
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s.RunFor(kRunTime);
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}
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// Using high error margin to avoid flakyness.
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const int kExpectedCount =
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static_cast<int>(kRunTime.seconds<double>() * kFrameRate * 0.5);
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EXPECT_GE(frame_counts[0], kExpectedCount);
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EXPECT_GE(frame_counts[1], kExpectedCount);
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EXPECT_GE(frame_counts[2], kExpectedCount);
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}
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TEST(VideoStreamTest, SendsNacksOnLoss) {
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Scenario s;
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auto route =
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s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
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{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
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c->loss_rate = 0.2;
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})},
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s.CreateClient("callee", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())});
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// NACK retransmissions are enabled by default.
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auto video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
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s.RunFor(TimeDelta::Seconds(1));
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int retransmit_packets = 0;
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for (const auto& substream : video->send()->GetStats().substreams) {
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retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
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}
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EXPECT_GT(retransmit_packets, 0);
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}
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TEST(VideoStreamTest, SendsFecWithUlpFec) {
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Scenario s;
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auto route =
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s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
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{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
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c->loss_rate = 0.1;
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c->delay = TimeDelta::Millis(100);
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})},
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s.CreateClient("callee", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())});
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auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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// We do not allow NACK+ULPFEC for generic codec, using VP8.
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c->encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
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c->stream.use_ulpfec = true;
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});
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s.RunFor(TimeDelta::Seconds(5));
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VideoSendStream::Stats video_stats = video->send()->GetStats();
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EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
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}
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TEST(VideoStreamTest, SendsFecWithFlexFec) {
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Scenario s;
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auto route =
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s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
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{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
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c->loss_rate = 0.1;
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c->delay = TimeDelta::Millis(100);
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})},
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s.CreateClient("callee", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())});
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auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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c->stream.use_flexfec = true;
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});
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s.RunFor(TimeDelta::Seconds(5));
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VideoSendStream::Stats video_stats = video->send()->GetStats();
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EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
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}
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TEST(VideoStreamTest, SendsFecWithDeferredFlexFec) {
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ScopedFieldTrials trial("WebRTC-DeferredFecGeneration/Enabled/");
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Scenario s;
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auto route =
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s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
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{s.CreateSimulationNode([](NetworkSimulationConfig* c) {
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c->loss_rate = 0.1;
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c->delay = TimeDelta::Millis(100);
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})},
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s.CreateClient("callee", CallClientConfig()),
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{s.CreateSimulationNode(NetworkSimulationConfig())});
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auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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c->stream.use_flexfec = true;
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});
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s.RunFor(TimeDelta::Seconds(5));
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VideoSendStream::Stats video_stats = video->send()->GetStats();
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EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
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}
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TEST(VideoStreamTest, ResolutionAdaptsToAvailableBandwidth) {
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// Declared before scenario to avoid use after free.
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std::atomic<size_t> num_qvga_frames_(0);
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std::atomic<size_t> num_vga_frames_(0);
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Scenario s;
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// Link has enough capacity for VGA.
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NetworkSimulationConfig net_conf;
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net_conf.bandwidth = DataRate::KilobitsPerSec(800);
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net_conf.delay = TimeDelta::Millis(50);
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auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
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c->transport.rates.start_rate = DataRate::KilobitsPerSec(800);
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});
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auto send_net = {s.CreateSimulationNode(net_conf)};
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auto ret_net = {s.CreateSimulationNode(net_conf)};
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auto* route = s.CreateRoutes(
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client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
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s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
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c->hooks.frame_pair_handlers = {[&](const VideoFramePair& info) {
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if (info.decoded->width() == 640) {
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++num_vga_frames_;
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} else if (info.decoded->width() == 320) {
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++num_qvga_frames_;
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} else {
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ADD_FAILURE() << "Unexpected resolution: " << info.decoded->width();
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}
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}};
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c->source.framerate = 30;
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// The resolution must be high enough to allow smaller layers to be
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// created.
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c->source.generator.width = 640;
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c->source.generator.height = 480;
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c->encoder.implementation = CodecImpl::kSoftware;
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c->encoder.codec = Codec::kVideoCodecVP9;
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// Enable SVC.
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c->encoder.layers.spatial = 2;
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});
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// Run for a few seconds, until streams have stabilized,
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// check that we are sending VGA.
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s.RunFor(TimeDelta::Seconds(5));
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EXPECT_GT(num_vga_frames_, 0u);
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// Trigger cross traffic, run until we have seen 3 consecutive
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// seconds with no VGA frames due to reduced available bandwidth.
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auto cross_traffic =
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s.net()->StartFakeTcpCrossTraffic(send_net, ret_net, FakeTcpConfig());
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int num_seconds_without_vga = 0;
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int num_iterations = 0;
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do {
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ASSERT_LE(++num_iterations, 100);
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num_qvga_frames_ = 0;
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num_vga_frames_ = 0;
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s.RunFor(TimeDelta::Seconds(1));
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if (num_qvga_frames_ > 0 && num_vga_frames_ == 0) {
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++num_seconds_without_vga;
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} else {
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num_seconds_without_vga = 0;
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}
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} while (num_seconds_without_vga < 3);
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// Stop cross traffic, make sure we recover and get VGA frames agian.
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s.net()->StopCrossTraffic(cross_traffic);
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num_qvga_frames_ = 0;
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num_vga_frames_ = 0;
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s.RunFor(TimeDelta::Seconds(40));
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EXPECT_GT(num_qvga_frames_, 0u);
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EXPECT_GT(num_vga_frames_, 0u);
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}
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} // namespace test
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} // namespace webrtc
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