webrtc_m130/test/scenario/video_stream_unittest.cc

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/*
* 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"
Reland "Reland "Allows FEC generation after pacer step."" This is a reland of 19df870d924662e3b6efb86078d31a8e086b38b5 Patchset 1 is the original. Subsequent patchset changes threadchecker that crashed with downstream code. Original change's description: > Reland "Allows FEC generation after pacer step." > > This is a reland of 75fd127640bdf1729af6b4a25875e6d01f1570e0 > > Patchset 2 contains a fix. Old code can in factor call > RtpRtcpImpl::FetchFec(). It should only be a noop since deferred fec > is not supported there - we shouldn't crash. > > Original change's description: > > Allows FEC generation after pacer step. > > > > Split out from https://webrtc-review.googlesource.com/c/src/+/173708 > > This CL enables FEC packets to be generated as media packets are sent, > > rather than generated, i.e. media packets are inserted into the fec > > generator after the pacing stage rather than at packetization time. > > > > This may have some small impact of performance. FEC packets are > > typically only generated when a new packet with a marker bit is added, > > which means FEC packets protecting a frame will now be sent after all > > of the media packets, rather than (potentially) interleaved with them. > > Therefore this feature is currently behind a flag so we can examine the > > impact. Once we are comfortable with the behavior we'll make it default > > and remove the old code. > > > > Note that this change does not include the "protect all header > > extensions" part of the original CL - that will be a follow-up. > > > > Bug: webrtc:11340 > > Change-Id: I3fe139c5d53968579b75b91e2612075451ff0f5d > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177760 > > Commit-Queue: Erik Språng <sprang@webrtc.org> > > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#31558} > > Bug: webrtc:11340 > Change-Id: I2ea49ee87ee9ff409044e34a777a7dd0ae0a077f > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/177984 > Commit-Queue: Erik Språng <sprang@webrtc.org> > Reviewed-by: Sebastian Jansson <srte@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#31613} Bug: webrtc:11340 Change-Id: Ib741c8c284f523c959f8aca454088d9eee7b17f8 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/178600 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Erik Språng <sprang@webrtc.org> Cr-Commit-Position: refs/heads/master@{#31619}
2020-07-02 17:41:32 +02:00
#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) {
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
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) {
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
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;
// 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());
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
s.RunFor(TimeDelta::Seconds(1));
int retransmit_packets = 0;
Reland "Remove Invoke from VideoChannel::FillBitrateInfo." This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb. Reason for reland: Relanding the original change but without the modification for VideoSendStream::GetStats. Essentially there's a TODO there to fix the downstream issue, which seems to be benign. Original change's description: > Revert "Remove Invoke from VideoChannel::FillBitrateInfo." > > This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14. > > Reason for revert: Speculative revert (breaks downstream project). > > Original change's description: > > Remove Invoke from VideoChannel::FillBitrateInfo. > > > > The method is relied upon by StatsCollector where it was called from the > > signaling thread in a loop. Now there's at most one invoke (not N). > > > > Uncommenting thread checks and removing TODOs in SendStatisticsProxy, > > VideoSendStream. Updating all related tests that fetched stats from > > the wrong context. > > > > Bug: webrtc:12726 > > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33894} > > TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I2520957cdb33492d187f04320c7416788fd0f820 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12726 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33898} # Not skipping CQ checks because this is a reland. Bug: webrtc:12726 Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33902}
2021-05-03 16:23:08 +02:00
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;
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
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;
});
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
s.RunFor(TimeDelta::Seconds(5));
Reland "Remove Invoke from VideoChannel::FillBitrateInfo." This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb. Reason for reland: Relanding the original change but without the modification for VideoSendStream::GetStats. Essentially there's a TODO there to fix the downstream issue, which seems to be benign. Original change's description: > Revert "Remove Invoke from VideoChannel::FillBitrateInfo." > > This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14. > > Reason for revert: Speculative revert (breaks downstream project). > > Original change's description: > > Remove Invoke from VideoChannel::FillBitrateInfo. > > > > The method is relied upon by StatsCollector where it was called from the > > signaling thread in a loop. Now there's at most one invoke (not N). > > > > Uncommenting thread checks and removing TODOs in SendStatisticsProxy, > > VideoSendStream. Updating all related tests that fetched stats from > > the wrong context. > > > > Bug: webrtc:12726 > > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33894} > > TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I2520957cdb33492d187f04320c7416788fd0f820 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12726 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33898} # Not skipping CQ checks because this is a reland. Bug: webrtc:12726 Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33902}
2021-05-03 16:23:08 +02:00
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;
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
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;
});
Use newer version of TimeDelta and TimeStamp factories in webrtc find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Micros<\(.*\)>()/TimeDelta::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Millis<\(.*\)>()/TimeDelta::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::Seconds<\(.*\)>()/TimeDelta::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::us/TimeDelta::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::ms/TimeDelta::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/TimeDelta::seconds/TimeDelta::Seconds/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Micros<\(.*\)>()/Timestamp::Micros(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Millis<\(.*\)>()/Timestamp::Millis(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::Seconds<\(.*\)>()/Timestamp::Seconds(\1)/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::us/Timestamp::Micros/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::ms/Timestamp::Millis/g" find . -type f \( -name "*.h" -o -name "*.cc" \) | xargs sed -i -e "s/Timestamp::seconds/Timestamp::Seconds/g" git cl format Bug: None Change-Id: I87469d2e4a38369654da839ab7c838215a7911e7 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/168402 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Commit-Queue: Danil Chapovalov <danilchap@webrtc.org> Cr-Commit-Position: refs/heads/master@{#30491}
2020-02-10 11:16:00 +01:00
s.RunFor(TimeDelta::Seconds(5));
Reland "Remove Invoke from VideoChannel::FillBitrateInfo." This reverts commit 48a4d33719390b7bcaf8445a1581a00825f67bfb. Reason for reland: Relanding the original change but without the modification for VideoSendStream::GetStats. Essentially there's a TODO there to fix the downstream issue, which seems to be benign. Original change's description: > Revert "Remove Invoke from VideoChannel::FillBitrateInfo." > > This reverts commit 1a1795768e1bdb65054ebe15aa238c6edc78dd14. > > Reason for revert: Speculative revert (breaks downstream project). > > Original change's description: > > Remove Invoke from VideoChannel::FillBitrateInfo. > > > > The method is relied upon by StatsCollector where it was called from the > > signaling thread in a loop. Now there's at most one invoke (not N). > > > > Uncommenting thread checks and removing TODOs in SendStatisticsProxy, > > VideoSendStream. Updating all related tests that fetched stats from > > the wrong context. > > > > Bug: webrtc:12726 > > Change-Id: Ia7db1afd7e103ec4f9816f5647203c4e2495586e > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/216688 > > Commit-Queue: Tommi <tommi@webrtc.org> > > Reviewed-by: Niels Moller <nisse@webrtc.org> > > Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> > > Cr-Commit-Position: refs/heads/master@{#33894} > > TBR=ilnik@webrtc.org,nisse@webrtc.org,tommi@webrtc.org,webrtc-scoped@luci-project-accounts.iam.gserviceaccount.com > > Change-Id: I2520957cdb33492d187f04320c7416788fd0f820 > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Bug: webrtc:12726 > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217240 > Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org> > Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org> > Cr-Commit-Position: refs/heads/master@{#33898} # Not skipping CQ checks because this is a reland. Bug: webrtc:12726 Change-Id: I41cce3b11a29905cde982c22e82b9b1f5a98e654 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217222 Reviewed-by: Tommi <tommi@webrtc.org> Reviewed-by: Niels Moller <nisse@webrtc.org> Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org> Commit-Queue: Tommi <tommi@webrtc.org> Cr-Commit-Position: refs/heads/master@{#33902}
2021-05-03 16:23:08 +02:00
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.layers.spatial = 2;
});
// 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