blob: 0789b0d5642074d0eca9ba49677c15667b103e94 [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 "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::ms(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, RecievesVp8SimulcastFrames) {
TimeDelta kRunTime = TimeDelta::ms(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;
// By enabling multiple spatial layers, simulcast will be enabled for VP8.
c->encoder.layers.spatial = 3;
});
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;
for (const auto& substream : video->send()->GetStats().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::ms(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 = 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::ms(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 = video->send()->GetStats();
EXPECT_GT(video_stats.substreams.begin()->second.rtp_stats.fec.packets, 0u);
}
} // namespace test
} // namespace webrtc