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authorDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
committerDaniel Baumann <daniel.baumann@progress-linux.org>2024-04-19 00:47:55 +0000
commit26a029d407be480d791972afb5975cf62c9360a6 (patch)
treef435a8308119effd964b339f76abb83a57c29483 /third_party/libwebrtc/test/scenario/video_stream_unittest.cc
parentInitial commit. (diff)
downloadfirefox-26a029d407be480d791972afb5975cf62c9360a6.tar.xz
firefox-26a029d407be480d791972afb5975cf62c9360a6.zip
Adding upstream version 124.0.1.upstream/124.0.1
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/test/scenario/video_stream_unittest.cc')
-rw-r--r--third_party/libwebrtc/test/scenario/video_stream_unittest.cc321
1 files changed, 321 insertions, 0 deletions
diff --git a/third_party/libwebrtc/test/scenario/video_stream_unittest.cc b/third_party/libwebrtc/test/scenario/video_stream_unittest.cc
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+++ b/third_party/libwebrtc/test/scenario/video_stream_unittest.cc
@@ -0,0 +1,321 @@
+/*
+ * Copyright 2019 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 <atomic>
+
+#include "api/test/network_emulation/create_cross_traffic.h"
+#include "api/test/network_emulation/cross_traffic.h"
+#include "test/field_trial.h"
+#include "test/gtest.h"
+#include "test/scenario/scenario.h"
+
+namespace webrtc {
+namespace test {
+namespace {
+using Capture = VideoStreamConfig::Source::Capture;
+using ContentType = VideoStreamConfig::Encoder::ContentType;
+using Codec = VideoStreamConfig::Encoder::Codec;
+using CodecImpl = VideoStreamConfig::Encoder::Implementation;
+} // namespace
+
+TEST(VideoStreamTest, ReceivesFramesFromFileBasedStreams) {
+ TimeDelta kRunTime = TimeDelta::Millis(500);
+ std::vector<int> kFrameRates = {15, 30};
+ std::deque<std::atomic<int>> frame_counts(2);
+ frame_counts[0] = 0;
+ frame_counts[1] = 0;
+ {
+ Scenario s;
+ auto route =
+ s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())},
+ s.CreateClient("callee", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())});
+
+ s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ c->hooks.frame_pair_handlers = {
+ [&](const VideoFramePair&) { frame_counts[0]++; }};
+ c->source.capture = Capture::kVideoFile;
+ c->source.video_file.name = "foreman_cif";
+ c->source.video_file.width = 352;
+ c->source.video_file.height = 288;
+ c->source.framerate = kFrameRates[0];
+ c->encoder.implementation = CodecImpl::kSoftware;
+ c->encoder.codec = Codec::kVideoCodecVP8;
+ });
+ s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ c->hooks.frame_pair_handlers = {
+ [&](const VideoFramePair&) { frame_counts[1]++; }};
+ c->source.capture = Capture::kImageSlides;
+ c->source.slides.images.crop.width = 320;
+ c->source.slides.images.crop.height = 240;
+ c->source.framerate = kFrameRates[1];
+ c->encoder.implementation = CodecImpl::kSoftware;
+ c->encoder.codec = Codec::kVideoCodecVP9;
+ });
+ s.RunFor(kRunTime);
+ }
+ std::vector<int> expected_counts;
+ for (int fps : kFrameRates)
+ expected_counts.push_back(
+ static_cast<int>(kRunTime.seconds<double>() * fps * 0.8));
+
+ EXPECT_GE(frame_counts[0], expected_counts[0]);
+ EXPECT_GE(frame_counts[1], expected_counts[1]);
+}
+
+TEST(VideoStreamTest, ReceivesVp8SimulcastFrames) {
+ TimeDelta kRunTime = TimeDelta::Millis(500);
+ int kFrameRate = 30;
+
+ std::deque<std::atomic<int>> frame_counts(3);
+ frame_counts[0] = 0;
+ frame_counts[1] = 0;
+ frame_counts[2] = 0;
+ {
+ Scenario s;
+ auto route =
+ s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())},
+ s.CreateClient("callee", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())});
+ s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ // TODO(srte): Replace with code checking for all simulcast streams when
+ // there's a hook available for that.
+ c->hooks.frame_pair_handlers = {[&](const VideoFramePair& info) {
+ frame_counts[info.layer_id]++;
+ RTC_DCHECK(info.decoded);
+ printf("%i: [%3i->%3i, %i], %i->%i, \n", info.layer_id, info.capture_id,
+ info.decode_id, info.repeated, info.captured->width(),
+ info.decoded->width());
+ }};
+ c->source.framerate = kFrameRate;
+ // The resolution must be high enough to allow smaller layers to be
+ // created.
+ c->source.generator.width = 1024;
+ c->source.generator.height = 768;
+ c->encoder.implementation = CodecImpl::kSoftware;
+ c->encoder.codec = Codec::kVideoCodecVP8;
+ // Enable simulcast.
+ c->encoder.simulcast_streams = {webrtc::ScalabilityMode::kL1T1,
+ webrtc::ScalabilityMode::kL1T1,
+ webrtc::ScalabilityMode::kL1T1};
+ });
+ s.RunFor(kRunTime);
+ }
+
+ // Using high error margin to avoid flakyness.
+ const int kExpectedCount =
+ static_cast<int>(kRunTime.seconds<double>() * kFrameRate * 0.5);
+
+ EXPECT_GE(frame_counts[0], kExpectedCount);
+ EXPECT_GE(frame_counts[1], kExpectedCount);
+ EXPECT_GE(frame_counts[2], kExpectedCount);
+}
+
+TEST(VideoStreamTest, SendsNacksOnLoss) {
+ Scenario s;
+ auto route =
+ s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+ {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+ c->loss_rate = 0.2;
+ })},
+ s.CreateClient("callee", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())});
+ // NACK retransmissions are enabled by default.
+ auto video = s.CreateVideoStream(route->forward(), VideoStreamConfig());
+ s.RunFor(TimeDelta::Seconds(1));
+ int retransmit_packets = 0;
+ VideoSendStream::Stats stats;
+ route->first()->SendTask([&]() { stats = video->send()->GetStats(); });
+ for (const auto& substream : stats.substreams) {
+ retransmit_packets += substream.second.rtp_stats.retransmitted.packets;
+ }
+ EXPECT_GT(retransmit_packets, 0);
+}
+
+TEST(VideoStreamTest, SendsFecWithUlpFec) {
+ Scenario s;
+ auto route =
+ s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+ {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+ c->loss_rate = 0.1;
+ c->delay = TimeDelta::Millis(100);
+ })},
+ s.CreateClient("callee", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())});
+ auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ // We do not allow NACK+ULPFEC for generic codec, using VP8.
+ c->encoder.codec = VideoStreamConfig::Encoder::Codec::kVideoCodecVP8;
+ c->stream.use_ulpfec = true;
+ });
+ s.RunFor(TimeDelta::Seconds(5));
+ VideoSendStream::Stats video_stats;
+ route->first()->SendTask([&]() { video_stats = video->send()->GetStats(); });
+ EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
+}
+TEST(VideoStreamTest, SendsFecWithFlexFec) {
+ Scenario s;
+ auto route =
+ s.CreateRoutes(s.CreateClient("caller", CallClientConfig()),
+ {s.CreateSimulationNode([](NetworkSimulationConfig* c) {
+ c->loss_rate = 0.1;
+ c->delay = TimeDelta::Millis(100);
+ })},
+ s.CreateClient("callee", CallClientConfig()),
+ {s.CreateSimulationNode(NetworkSimulationConfig())});
+ auto video = s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ c->stream.use_flexfec = true;
+ });
+ s.RunFor(TimeDelta::Seconds(5));
+ VideoSendStream::Stats video_stats;
+ route->first()->SendTask([&]() { video_stats = video->send()->GetStats(); });
+ EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
+}
+
+TEST(VideoStreamTest, ResolutionAdaptsToAvailableBandwidth) {
+ // Declared before scenario to avoid use after free.
+ std::atomic<size_t> num_qvga_frames_(0);
+ std::atomic<size_t> num_vga_frames_(0);
+
+ Scenario s;
+ // Link has enough capacity for VGA.
+ NetworkSimulationConfig net_conf;
+ net_conf.bandwidth = DataRate::KilobitsPerSec(800);
+ net_conf.delay = TimeDelta::Millis(50);
+ auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+ c->transport.rates.start_rate = DataRate::KilobitsPerSec(800);
+ });
+ auto send_net = {s.CreateSimulationNode(net_conf)};
+ auto ret_net = {s.CreateSimulationNode(net_conf)};
+ auto* route = s.CreateRoutes(
+ client, send_net, s.CreateClient("return", CallClientConfig()), ret_net);
+
+ s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ c->hooks.frame_pair_handlers = {[&](const VideoFramePair& info) {
+ if (info.decoded->width() == 640) {
+ ++num_vga_frames_;
+ } else if (info.decoded->width() == 320) {
+ ++num_qvga_frames_;
+ } else {
+ ADD_FAILURE() << "Unexpected resolution: " << info.decoded->width();
+ }
+ }};
+ c->source.framerate = 30;
+ // The resolution must be high enough to allow smaller layers to be
+ // created.
+ c->source.generator.width = 640;
+ c->source.generator.height = 480;
+ c->encoder.implementation = CodecImpl::kSoftware;
+ c->encoder.codec = Codec::kVideoCodecVP9;
+ // Enable SVC.
+ c->encoder.simulcast_streams = {webrtc::ScalabilityMode::kL2T1};
+ });
+
+ // Run for a few seconds, until streams have stabilized,
+ // check that we are sending VGA.
+ s.RunFor(TimeDelta::Seconds(5));
+ EXPECT_GT(num_vga_frames_, 0u);
+
+ // Trigger cross traffic, run until we have seen 3 consecutive
+ // seconds with no VGA frames due to reduced available bandwidth.
+ auto cross_traffic = s.net()->StartCrossTraffic(CreateFakeTcpCrossTraffic(
+ s.net()->CreateRoute(send_net), s.net()->CreateRoute(ret_net),
+ FakeTcpConfig()));
+
+ int num_seconds_without_vga = 0;
+ int num_iterations = 0;
+ do {
+ ASSERT_LE(++num_iterations, 100);
+ num_qvga_frames_ = 0;
+ num_vga_frames_ = 0;
+ s.RunFor(TimeDelta::Seconds(1));
+ if (num_qvga_frames_ > 0 && num_vga_frames_ == 0) {
+ ++num_seconds_without_vga;
+ } else {
+ num_seconds_without_vga = 0;
+ }
+ } while (num_seconds_without_vga < 3);
+
+ // Stop cross traffic, make sure we recover and get VGA frames agian.
+ s.net()->StopCrossTraffic(cross_traffic);
+ num_qvga_frames_ = 0;
+ num_vga_frames_ = 0;
+
+ s.RunFor(TimeDelta::Seconds(40));
+ EXPECT_GT(num_qvga_frames_, 0u);
+ EXPECT_GT(num_vga_frames_, 0u);
+}
+
+TEST(VideoStreamTest, SuspendsBelowMinBitrate) {
+ const DataRate kMinVideoBitrate = DataRate::KilobitsPerSec(30);
+
+ // Declared before scenario to avoid use after free.
+ std::atomic<Timestamp> last_frame_timestamp(Timestamp::MinusInfinity());
+
+ Scenario s;
+ NetworkSimulationConfig net_config;
+ net_config.bandwidth = kMinVideoBitrate * 4;
+ net_config.delay = TimeDelta::Millis(10);
+ auto* client = s.CreateClient("send", [&](CallClientConfig* c) {
+ // Min transmit rate needs to be lower than kMinVideoBitrate for this test
+ // to make sense.
+ c->transport.rates.min_rate = kMinVideoBitrate / 2;
+ c->transport.rates.start_rate = kMinVideoBitrate;
+ c->transport.rates.max_rate = kMinVideoBitrate * 2;
+ });
+ auto send_net = s.CreateMutableSimulationNode(
+ [&](NetworkSimulationConfig* c) { *c = net_config; });
+ auto ret_net = {s.CreateSimulationNode(net_config)};
+ auto* route =
+ s.CreateRoutes(client, {send_net->node()},
+ s.CreateClient("return", CallClientConfig()), ret_net);
+
+ s.CreateVideoStream(route->forward(), [&](VideoStreamConfig* c) {
+ c->hooks.frame_pair_handlers = {[&](const VideoFramePair& pair) {
+ if (pair.repeated == 0) {
+ last_frame_timestamp = pair.capture_time;
+ }
+ }};
+ c->source.framerate = 30;
+ c->source.generator.width = 320;
+ c->source.generator.height = 180;
+ c->encoder.implementation = CodecImpl::kFake;
+ c->encoder.codec = Codec::kVideoCodecVP8;
+ c->encoder.min_data_rate = kMinVideoBitrate;
+ c->encoder.suspend_below_min_bitrate = true;
+ c->stream.pad_to_rate = kMinVideoBitrate;
+ });
+
+ // Run for a few seconds, check we have received at least one frame.
+ s.RunFor(TimeDelta::Seconds(2));
+ EXPECT_TRUE(last_frame_timestamp.load().IsFinite());
+
+ // Degrade network to below min bitrate.
+ send_net->UpdateConfig([&](NetworkSimulationConfig* c) {
+ c->bandwidth = kMinVideoBitrate * 0.9;
+ });
+
+ // Run for 20s, verify that no frames arrive that were captured after the
+ // first five seconds, allowing some margin for BWE backoff to trigger and
+ // packets already in the pipeline to potentially arrive.
+ s.RunFor(TimeDelta::Seconds(20));
+ EXPECT_GT(s.Now() - last_frame_timestamp, TimeDelta::Seconds(15));
+
+ // Relax the network constraints and run for a while more, verify that we
+ // start receiving frames again.
+ send_net->UpdateConfig(
+ [&](NetworkSimulationConfig* c) { c->bandwidth = kMinVideoBitrate * 4; });
+ last_frame_timestamp = Timestamp::MinusInfinity();
+ s.RunFor(TimeDelta::Seconds(15));
+ EXPECT_TRUE(last_frame_timestamp.load().IsFinite());
+}
+
+} // namespace test
+} // namespace webrtc