| /* |
| * 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 |