blob: 3e89c785560f6390625b66507bcba95bd2585485 [file] [log] [blame]
/*
* 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);
#ifndef __ANDROID__
// TODO: crbug.com/webrtc/15873 - This expectation is flaky on Android.
EXPECT_GT(num_vga_frames_, 0u);
#endif
}
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