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Use GlobalSimulatedTimeController in more webrtc video engine unittests

Browse source 
This removes all "_WAIT" operations in the tests and all uses
of kTimeout for loop+poll check for values.

On my linux box, this also reduces the time it takes to run all
./rtc_media_unittests (in parallel) from about 3500ms, to ~50ms.
(longest running test was WebRtcVideoChannelBaseTest.GetStats)

Bug: none
Change-Id: If544aa10cb0281cb5e5e51fb654db5f45de871da
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/302343
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#39918}
This commit is contained in:
Tommi 2023-04-21 16:07:42 +02:00 committed by WebRTC LUCI CQ
parent 691b447c53
commit 51c632c13b
1 changed files with 109 additions and 121 deletions
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230
media/engine/webrtc_video_engine_unittest.cc
View file

@ -103,6 +103,8 @@ using ::webrtc::RtpExtension;
using ::webrtc::RtpPacket;
using ::webrtc::RtpPacketReceived;
using ::webrtc::ScalabilityMode;
using ::webrtc::TimeDelta;
using ::webrtc::Timestamp;
using ::webrtc::test::RtcpPacketParser;
namespace {
@ -110,12 +112,13 @@ static const int kDefaultQpMax = 56;
static const uint8_t kRedRtxPayloadType = 125;
static const uint32_t kTimeout = 5000U;
static const uint32_t kSsrc = 1234u;
static const uint32_t kSsrcs4[] = {1, 2, 3, 4};
static const int kVideoWidth = 640;
static const int kVideoHeight = 360;
static const int kFramerate = 30;
static constexpr TimeDelta kFrameDuration =
TimeDelta::Millis(1000 / kFramerate);
static const uint32_t kSsrcs1[] = {1};
static const uint32_t kSsrcs3[] = {1, 2, 3};
@ -332,14 +335,15 @@ RtpPacketReceived BuildRtxPacket(uint32_t rtx_ssrc,
} // namespace
#define EXPECT_FRAME_WAIT(c, w, h, t) \
EXPECT_EQ_WAIT((c), renderer_.num_rendered_frames(), (t)); \
EXPECT_EQ((w), renderer_.width()); \
// TODO(tommi): Consider replacing these macros with custom matchers.
#define EXPECT_FRAME(c, w, h) \
EXPECT_EQ((c), renderer_.num_rendered_frames()); \
EXPECT_EQ((w), renderer_.width()); \
EXPECT_EQ((h), renderer_.height());
#define EXPECT_FRAME_ON_RENDERER_WAIT(r, c, w, h, t) \
EXPECT_EQ_WAIT((c), (r).num_rendered_frames(), (t)); \
EXPECT_EQ((w), (r).width()); \
#define EXPECT_FRAME_ON_RENDERER(r, c, w, h) \
EXPECT_EQ((c), (r).num_rendered_frames()); \
EXPECT_EQ((w), (r).width()); \
EXPECT_EQ((h), (r).height());
namespace cricket {
@ -671,8 +675,7 @@ TEST_F(WebRtcVideoEngineTest, UseFactoryForVp8WhenSupported) {
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
// Sending one frame will have allocate the encoder.
ASSERT_TRUE(encoder_factory_->WaitForCreatedVideoEncoders(1));
EXPECT_TRUE_WAIT(encoder_factory_->encoders()[0]->GetNumEncodedFrames() > 0,
kTimeout);
EXPECT_GT(encoder_factory_->encoders()[0]->GetNumEncodedFrames(), 0);
int num_created_encoders = encoder_factory_->GetNumCreatedEncoders();
EXPECT_EQ(num_created_encoders, 1);
@ -1061,7 +1064,7 @@ TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) {
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
ASSERT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
ASSERT_EQ(1u, encoder_factory_->encoders().size());
cricket::VideoSendParameters new_parameters;
new_parameters.codecs.push_back(GetEngineCodec("VP8"));
@ -1071,7 +1074,7 @@ TEST_F(WebRtcVideoEngineTest, ChannelWithH264CanChangeToVp8) {
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
time_controller_.AdvanceTime(webrtc::TimeDelta::Zero());
EXPECT_EQ_WAIT(1u, encoder_factory_->encoders().size(), kTimeout);
EXPECT_EQ(1u, encoder_factory_->encoders().size());
}
TEST_F(WebRtcVideoEngineTest,
@ -1595,7 +1598,7 @@ class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
}
WebRtcVideoChannelEncodedFrameCallbackTest()
: task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
: task_queue_factory_(time_controller_.CreateTaskQueueFactory()),
call_(absl::WrapUnique(webrtc::Call::Create(
GetCallConfig(&event_log_, task_queue_factory_.get())))),
video_bitrate_allocator_factory_(
@ -1639,11 +1642,13 @@ class WebRtcVideoChannelEncodedFrameCallbackTest : public ::testing::Test {
void DeliverKeyFrameAndWait(uint32_t ssrc) {
DeliverKeyFrame(ssrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_EQ(renderer_.num_rendered_frames(), 1);
}
static const std::vector<webrtc::SdpVideoFormat> kSdpVideoFormats;
rtc::AutoThread main_thread_;
webrtc::GlobalSimulatedTimeController time_controller_{
Timestamp::Seconds(1000)};
webrtc::test::ScopedKeyValueConfig field_trials_;
webrtc::RtcEventLogNull event_log_;
std::unique_ptr<webrtc::TaskQueueFactory> task_queue_factory_;
@ -1672,7 +1677,8 @@ TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
callback.AsStdFunction());
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
DeliverKeyFrame(kSsrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_EQ(renderer_.num_rendered_frames(), 1);
receive_channel_->RemoveRecvStream(kSsrc);
}
@ -1685,7 +1691,8 @@ TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction());
DeliverKeyFrame(kSsrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_EQ(renderer_.num_rendered_frames(), 1);
receive_channel_->RemoveRecvStream(kSsrc);
}
@ -1698,7 +1705,8 @@ TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
channel_->SetRecordableEncodedFrameCallback(kSsrc, callback.AsStdFunction());
DeliverKeyFrame(kSsrc);
EXPECT_EQ_WAIT(1, renderer_.num_rendered_frames(), kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_EQ(renderer_.num_rendered_frames(), 1);
receive_channel_->RemoveRecvStream(kSsrc);
}
@ -1736,7 +1744,7 @@ TEST_F(WebRtcVideoChannelEncodedFrameCallbackTest,
class WebRtcVideoChannelBaseTest : public ::testing::Test {
protected:
WebRtcVideoChannelBaseTest()
: task_queue_factory_(webrtc::CreateDefaultTaskQueueFactory()),
: task_queue_factory_(time_controller_.CreateTaskQueueFactory()),
video_bitrate_allocator_factory_(
webrtc::CreateBuiltinVideoBitrateAllocatorFactory()),
engine_(std::make_unique<webrtc::VideoEncoderFactoryTemplate<
@ -1759,6 +1767,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
call_config.trials = &field_trials_;
call_.reset(webrtc::Call::Create(call_config));
}
cricket::MediaConfig media_config;
// Disabling cpu overuse detection actually disables quality scaling too; it
// implies DegradationPreference kMaintainResolution. Automatic scaling
@ -1851,11 +1860,12 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
frame_forwarder_2_->IncomingCapturedFrame(frame_source_->GetFrame());
}
frame_forwarder_->IncomingCapturedFrame(frame_source_->GetFrame());
time_controller_.AdvanceTime(kFrameDuration);
}
bool WaitAndSendFrame(int wait_ms) {
bool ret = rtc::Thread::Current()->ProcessMessages(wait_ms);
time_controller_.AdvanceTime(TimeDelta::Millis(wait_ms));
SendFrame();
return ret;
return true;
}
int NumRtpBytes() { return network_interface_.NumRtpBytes(); }
int NumRtpBytes(uint32_t ssrc) {
@ -1882,7 +1892,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
EXPECT_EQ(codec.id, GetPayloadType(GetRtpPacket(0)));
}
@ -1896,7 +1906,7 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
for (int i = 0; i < duration_sec; ++i) {
for (int frame = 1; frame <= fps; ++frame) {
EXPECT_TRUE(WaitAndSendFrame(1000 / fps));
EXPECT_FRAME_WAIT(frame + i * fps, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(frame + i * fps, kVideoWidth, kVideoHeight);
}
}
EXPECT_EQ(codec.id, GetPayloadType(GetRtpPacket(0)));
@ -1922,9 +1932,8 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
// Test sending and receiving on first stream.
SendAndReceive(codec);
// Test sending and receiving on second stream.
EXPECT_EQ_WAIT(1, renderer2_.num_rendered_frames(), kTimeout);
EXPECT_EQ(renderer2_.num_rendered_frames(), 1);
EXPECT_GT(NumRtpPackets(), 0);
EXPECT_EQ(1, renderer2_.num_rendered_frames());
}
cricket::VideoCodec GetEngineCodec(const std::string& name) {
@ -1943,7 +1952,9 @@ class WebRtcVideoChannelBaseTest : public ::testing::Test {
return cricket::StreamParams::CreateLegacy(kSsrc);
}
rtc::AutoThread main_thread_;
webrtc::GlobalSimulatedTimeController time_controller_{
Timestamp::Seconds(1000)};
webrtc::RtcEventLogNull event_log_;
webrtc::test::ScopedKeyValueConfig field_trials_;
std::unique_ptr<webrtc::test::ScopedKeyValueConfig> override_field_trials_;
@ -1976,7 +1987,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSend) {
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(channel_->sending());
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
EXPECT_TRUE(SetSend(false));
EXPECT_FALSE(channel_->sending());
}
@ -2078,10 +2089,8 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
ssrcs.push_back(2);
network_interface_.SetConferenceMode(true, ssrcs);
SendFrame();
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER(renderer1, 1, kVideoWidth, kVideoHeight);
EXPECT_FRAME_ON_RENDERER(renderer2, 1, kVideoWidth, kVideoHeight);
EXPECT_TRUE(channel_->SetSend(false));
@ -2093,9 +2102,9 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
ASSERT_EQ(1U, send_info.senders.size());
// TODO(whyuan): bytes_sent and bytes_rcvd are different. Are both payload?
// For webrtc, bytes_sent does not include the RTP header length.
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetSenderStats(0).payload_bytes_sent, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetSenderStats(0).packets_sent, kTimeout);
EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetSenderStats(0).payload_bytes_sent);
EXPECT_EQ(NumRtpPackets(), GetSenderStats(0).packets_sent);
EXPECT_EQ(kVideoWidth, GetSenderStats(0).send_frame_width);
EXPECT_EQ(kVideoHeight, GetSenderStats(0).send_frame_height);
@ -2103,11 +2112,11 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
for (size_t i = 0; i < receive_info.receivers.size(); ++i) {
EXPECT_EQ(1U, GetReceiverStats(i).ssrcs().size());
EXPECT_EQ(i + 1, GetReceiverStats(i).ssrcs()[0]);
EXPECT_EQ_WAIT(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetReceiverStats(i).payload_bytes_rcvd, kTimeout);
EXPECT_EQ_WAIT(NumRtpPackets(), GetReceiverStats(i).packets_rcvd, kTimeout);
EXPECT_EQ_WAIT(kVideoWidth, GetReceiverStats(i).frame_width, kTimeout);
EXPECT_EQ_WAIT(kVideoHeight, GetReceiverStats(i).frame_height, kTimeout);
EXPECT_EQ(NumRtpBytes() - kRtpHeaderSize * NumRtpPackets(),
GetReceiverStats(i).payload_bytes_rcvd);
EXPECT_EQ(NumRtpPackets(), GetReceiverStats(i).packets_rcvd);
EXPECT_EQ(kVideoWidth, GetReceiverStats(i).frame_width);
EXPECT_EQ(kVideoHeight, GetReceiverStats(i).frame_height);
}
}
@ -2125,8 +2134,8 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) {
EXPECT_TRUE(channel_->SetSink(kSsrc, &renderer_));
EXPECT_TRUE(SetSend(true));
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
// Add an additional capturer, and hook up a renderer to receive it.
cricket::FakeVideoRenderer renderer2;
@ -2142,28 +2151,17 @@ TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleSendStreams) {
cricket::StreamParams::CreateLegacy(5678)));
EXPECT_TRUE(channel_->SetSink(5678, &renderer2));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_FRAME_ON_RENDERER(renderer2, 1, kTestWidth, kTestHeight);
// Get stats, and make sure they are correct for two senders. We wait until
// the number of expected packets have been sent to avoid races where we
// check stats before it has been updated.
// Get stats, and make sure they are correct for two senders
cricket::VideoMediaSendInfo send_info;
for (uint32_t i = 0; i < kTimeout; ++i) {
rtc::Thread::Current()->ProcessMessages(1);
EXPECT_TRUE(channel_->GetSendStats(&send_info));
EXPECT_TRUE(channel_->GetSendStats(&send_info));
ASSERT_EQ(2U, send_info.senders.size());
ASSERT_EQ(2U, send_info.senders.size());
if (send_info.senders[0].packets_sent + send_info.senders[1].packets_sent ==
NumRtpPackets()) {
// Stats have been updated for both sent frames, expectations can be
// checked now.
break;
}
}
EXPECT_EQ(NumRtpPackets(), send_info.senders[0].packets_sent +
send_info.senders[1].packets_sent)
<< "Timed out while waiting for packet counts for all sent packets.";
send_info.senders[1].packets_sent);
EXPECT_EQ(1U, send_info.senders[0].ssrcs().size());
EXPECT_EQ(1234U, send_info.senders[0].ssrcs()[0]);
EXPECT_EQ(kVideoWidth, send_info.senders[0].send_frame_width);
@ -2191,15 +2189,15 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrc) {
EXPECT_TRUE(SetDefaultCodec());
EXPECT_TRUE(SetSend(true));
SendFrame();
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
RtpPacket header;
EXPECT_TRUE(header.Parse(GetRtpPacket(0)));
EXPECT_EQ(kSsrc, header.Ssrc());
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.Ssrc()), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.Ssrc()), kTimeout);
EXPECT_EQ(NumRtpPackets(), NumRtpPackets(header.Ssrc()));
EXPECT_EQ(NumRtpBytes(), NumRtpBytes(header.Ssrc()));
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc - 1));
EXPECT_EQ(0, NumRtpBytes(kSsrc - 1));
@ -2215,14 +2213,14 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSendSsrcAfterSetCodecs) {
EXPECT_TRUE(channel_->SetVideoSend(999u, nullptr, frame_forwarder_.get()));
EXPECT_TRUE(SetSend(true));
EXPECT_TRUE(WaitAndSendFrame(0));
EXPECT_TRUE_WAIT(NumRtpPackets() > 0, kTimeout);
EXPECT_GT(NumRtpPackets(), 0);
RtpPacket header;
EXPECT_TRUE(header.Parse(GetRtpPacket(0)));
EXPECT_EQ(999u, header.Ssrc());
// Packets are being paced out, so these can mismatch between the first and
// second call to NumRtpPackets until pending packets are paced out.
EXPECT_EQ_WAIT(NumRtpPackets(), NumRtpPackets(header.Ssrc()), kTimeout);
EXPECT_EQ_WAIT(NumRtpBytes(), NumRtpBytes(header.Ssrc()), kTimeout);
EXPECT_EQ(NumRtpPackets(), NumRtpPackets(header.Ssrc()));
EXPECT_EQ(NumRtpBytes(), NumRtpBytes(header.Ssrc()));
EXPECT_EQ(1, NumSentSsrcs());
EXPECT_EQ(0, NumRtpPackets(kSsrc));
EXPECT_EQ(0, NumRtpBytes(kSsrc));
@ -2240,7 +2238,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SetSink) {
channel_->SetDefaultSink(&renderer_);
channel_->OnPacketReceived(packet);
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
}
// Tests setting up and configuring a send stream.
@ -2249,7 +2247,7 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
EXPECT_TRUE(SetSend(true));
channel_->SetDefaultSink(&renderer_);
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
EXPECT_GT(NumRtpPackets(), 0);
RtpPacket header;
size_t last_packet = NumRtpPackets() - 1;
@ -2266,7 +2264,7 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveSendStreams) {
EXPECT_EQ(rtp_packets, NumRtpPackets());
// Wait 30ms to guarantee the engine does not drop the frame.
EXPECT_TRUE(WaitAndSendFrame(30));
EXPECT_TRUE_WAIT(NumRtpPackets() > rtp_packets, kTimeout);
EXPECT_GT(NumRtpPackets(), rtp_packets);
last_packet = NumRtpPackets() - 1;
EXPECT_TRUE(header.Parse(GetRtpPacket(static_cast<int>(last_packet))));
@ -2295,10 +2293,8 @@ TEST_F(WebRtcVideoChannelBaseTest, SimulateConference) {
ssrcs.push_back(2);
network_interface_.SetConferenceMode(true, ssrcs);
SendFrame();
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kVideoWidth, kVideoHeight,
kTimeout);
EXPECT_FRAME_ON_RENDERER(renderer1, 1, kVideoWidth, kVideoHeight);
EXPECT_FRAME_ON_RENDERER(renderer2, 1, kVideoWidth, kVideoHeight);
EXPECT_EQ(DefaultCodec().id, GetPayloadType(GetRtpPacket(0)));
EXPECT_EQ(kVideoWidth, renderer1.width());
@ -2323,7 +2319,7 @@ TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) {
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
webrtc::test::FrameForwarder frame_forwarder;
cricket::FakeFrameSource frame_source(480, 360, rtc::kNumMicrosecsPerSec / 30,
@ -2339,28 +2335,26 @@ TEST_F(WebRtcVideoChannelBaseTest, DISABLED_AddRemoveCapturer) {
int captured_frames = 1;
for (int iterations = 0; iterations < 2; ++iterations) {
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, &frame_forwarder));
rtc::Thread::Current()->ProcessMessages(time_between_send_ms);
time_controller_.AdvanceTime(TimeDelta::Millis(time_between_send_ms));
frame_forwarder.IncomingCapturedFrame(frame_source.GetFrame());
++captured_frames;
// Wait until frame of right size is captured.
EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() &&
360 == renderer_.height() && !renderer_.black_frame(),
kTimeout);
// Check if the right size was captured.
EXPECT_TRUE(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() && 360 == renderer_.height() &&
!renderer_.black_frame());
EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
EXPECT_EQ(480, renderer_.width());
EXPECT_EQ(360, renderer_.height());
captured_frames = renderer_.num_rendered_frames() + 1;
EXPECT_FALSE(renderer_.black_frame());
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// Make sure a black frame is generated within the specified timeout.
// The black frame should be the resolution of the previous frame to
// Make sure a black frame was generated.
// The black frame should have the resolution of the previous frame to
// prevent expensive encoder reconfigurations.
EXPECT_TRUE_WAIT(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() &&
360 == renderer_.height() && renderer_.black_frame(),
kTimeout);
EXPECT_TRUE(renderer_.num_rendered_frames() >= captured_frames &&
480 == renderer_.width() && 360 == renderer_.height() &&
renderer_.black_frame());
EXPECT_GE(renderer_.num_rendered_frames(), captured_frames);
EXPECT_EQ(480, renderer_.width());
EXPECT_EQ(360, renderer_.height());
@ -2383,26 +2377,22 @@ TEST_F(WebRtcVideoChannelBaseTest, RemoveCapturerWithoutAdd) {
channel_->SetDefaultSink(&renderer_);
EXPECT_EQ(0, renderer_.num_rendered_frames());
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
// Wait for one frame so they don't get dropped because we send frames too
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
// Allow one frame so they don't get dropped because we send frames too
// tightly.
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
// Remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// No capturer was added, so this SetVideoSend shouldn't do anything.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
rtc::Thread::Current()->ProcessMessages(300);
time_controller_.AdvanceTime(TimeDelta::Millis(300));
// Verify no more frames were sent.
EXPECT_EQ(1, renderer_.num_rendered_frames());
}
// Tests that we can add and remove capturer as unique sources.
TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) {
// WebRTC implementation will drop frames if pushed to quickly. Wait the
// interval time to avoid that.
// WebRTC implementation will drop frames if pushed to quickly. Wait the
// interval time to avoid that.
// Set up the stream associated with the engine.
EXPECT_TRUE(receive_channel_->AddRecvStream(
cricket::StreamParams::CreateLegacy(kSsrc)));
@ -2447,14 +2437,14 @@ TEST_F(WebRtcVideoChannelBaseTest, AddRemoveCapturerMultipleSources) {
frame_forwarder1.IncomingCapturedFrame(frame_source.GetFrame(
kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / kFramerate));
EXPECT_FRAME_ON_RENDERER_WAIT(renderer1, 1, kTestWidth, kTestHeight,
kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_FRAME_ON_RENDERER(renderer1, 1, kTestWidth, kTestHeight);
// Capture a frame with additional capturer2, frames should be received
frame_forwarder2.IncomingCapturedFrame(frame_source.GetFrame(
kTestWidth, kTestHeight, webrtc::VideoRotation::kVideoRotation_0,
rtc::kNumMicrosecsPerSec / kFramerate));
EXPECT_FRAME_ON_RENDERER_WAIT(renderer2, 1, kTestWidth, kTestHeight,
kTimeout);
time_controller_.AdvanceTime(kFrameDuration);
EXPECT_FRAME_ON_RENDERER(renderer2, 1, kTestWidth, kTestHeight);
// Successfully remove the capturer.
EXPECT_TRUE(channel_->SetVideoSend(kSsrc, nullptr, nullptr));
// The capturers must be unregistered here as it runs out of it's scope
@ -2529,13 +2519,13 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderFallback) {
// RequestEncoderFallback will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP8", codec.name);
// No other codec to fall back to, keep using VP8.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP8", codec.name);
}
@ -2554,7 +2544,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchDefaultFallback) {
// the current thread), hence the ProcessMessages call.
channel_->RequestEncoderSwitch(webrtc::SdpVideoFormat("UnavailableCodec"),
/*allow_default_fallback=*/true);
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
// Requested encoder is not available. Default fallback is allowed. Switch to
// the next negotiated codec, VP8.
@ -2578,7 +2568,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchStrictPreference) {
channel_->RequestEncoderSwitch(
webrtc::SdpVideoFormat("VP9", {{"profile-id", "1"}}),
/*allow_default_fallback=*/false);
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
// VP9 profile_id=1 is not available. Default fallback is not allowed. Switch
// is not performed.
@ -2588,7 +2578,7 @@ TEST_F(WebRtcVideoChannelBaseTest, RequestEncoderSwitchStrictPreference) {
channel_->RequestEncoderSwitch(
webrtc::SdpVideoFormat("VP9", {{"profile-id", "0"}}),
/*allow_default_fallback=*/false);
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
// VP9 profile_id=0 is available. Switch encoder.
ASSERT_TRUE(channel_->GetSendCodec(&codec));
@ -2609,7 +2599,7 @@ TEST_F(WebRtcVideoChannelBaseTest, SendCodecIsMovedToFrontInRtpParameters) {
// RequestEncoderFallback will post a task to the worker thread (which is also
// the current thread), hence the ProcessMessages call.
channel_->RequestEncoderFallback();
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
send_codecs = send_channel_->GetRtpSendParameters(kSsrc).codecs;
ASSERT_EQ(send_codecs.size(), 2u);
@ -2675,7 +2665,6 @@ class WebRtcVideoChannelTest : public WebRtcVideoEngineTest {
// the unsignalled receive stream cooldown is no longer in effect.
void ReceivePacketAndAdvanceTime(const RtpPacketReceived& packet) {
receive_channel_->OnPacketReceived(packet);
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(
webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs));
}
@ -3252,7 +3241,7 @@ TEST_F(WebRtcVideoChannelTest, OnPacketReceivedIdentifiesExtensions) {
ASSERT_TRUE(received_packet.Parse(reference_packet.Buffer()));
receive_channel_->OnPacketReceived(received_packet);
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
EXPECT_EQ(fake_call_->last_received_rtp_packet()
.GetExtension<webrtc::VideoOrientation>(),
@ -6873,7 +6862,7 @@ TEST_F(WebRtcVideoChannelTest, RecvUnsignaledSsrcWithSignaledStreamId) {
ASSERT_TRUE(receive_channel_->AddRecvStream(unsignaled_stream));
receive_channel_->OnDemuxerCriteriaUpdatePending();
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// The stream shouldn't have been created at this point because it doesn't
// have any SSRCs.
EXPECT_EQ(0u, fake_call_->GetVideoReceiveStreams().size());
@ -6947,7 +6936,7 @@ TEST_F(WebRtcVideoChannelTest,
receive_channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc1)));
receive_channel_->OnDemuxerCriteriaUpdatePending();
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
// If this is the only m= section the demuxer might be configure to forward
@ -6984,7 +6973,7 @@ TEST_F(WebRtcVideoChannelTest,
// pending demuxer updates, receiving unknown ssrcs (kSsrc2) should again
// result in unsignalled receive streams being created.
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// Receive packets for kSsrc1 and kSsrc2 again.
{
@ -7019,7 +7008,7 @@ TEST_F(WebRtcVideoChannelTest,
receive_channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc2)));
receive_channel_->OnDemuxerCriteriaUpdatePending();
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
EXPECT_EQ(fake_call_->GetVideoReceiveStreams().size(), 2u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc1), 0u);
EXPECT_EQ(fake_call_->GetDeliveredPacketsForSsrc(kSsrc2), 0u);
@ -7056,7 +7045,7 @@ TEST_F(WebRtcVideoChannelTest,
// Signal that the demuxer update is complete. This means we should stop
// ignorning kSsrc1.
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// Receive packets for kSsrc1 and kSsrc2 again.
{
@ -7087,7 +7076,7 @@ TEST_F(WebRtcVideoChannelTest, MultiplePendingDemuxerCriteriaUpdates) {
receive_channel_->AddRecvStream(StreamParams::CreateLegacy(kSsrc)));
receive_channel_->OnDemuxerCriteriaUpdatePending();
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
ASSERT_EQ(fake_call_->GetVideoReceiveStreams().size(), 1u);
// Remove kSsrc...
@ -7112,7 +7101,7 @@ TEST_F(WebRtcVideoChannelTest, MultiplePendingDemuxerCriteriaUpdates) {
// Signal that the demuxer knows about the first update: the removal.
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// This still should not prevent in-flight packets from arriving because we
// have a receive stream for it.
@ -7140,7 +7129,7 @@ TEST_F(WebRtcVideoChannelTest, MultiplePendingDemuxerCriteriaUpdates) {
// Signal that the demuxer knows about the second update: adding it back.
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// The packets should continue to be dropped because removal happened after
// the most recently completed demuxer update.
@ -7154,7 +7143,7 @@ TEST_F(WebRtcVideoChannelTest, MultiplePendingDemuxerCriteriaUpdates) {
// Signal that the demuxer knows about the last update: the second removal.
receive_channel_->OnDemuxerCriteriaUpdateComplete();
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// If packets still arrive after the demuxer knows about the latest removal we
// should finally create an unsignalled receive stream.
@ -7178,7 +7167,7 @@ TEST_F(WebRtcVideoChannelTest, UnsignalledSsrcHasACooldown) {
packet.SetSsrc(kSsrc1);
receive_channel_->OnPacketReceived(packet);
}
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(
webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs - 1));
@ -7193,7 +7182,7 @@ TEST_F(WebRtcVideoChannelTest, UnsignalledSsrcHasACooldown) {
packet.SetSsrc(kSsrc2);
receive_channel_->OnPacketReceived(packet);
}
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// Not enough time has passed to replace the unsignalled receive stream, so
// the kSsrc2 should be ignored.
@ -7210,7 +7199,7 @@ TEST_F(WebRtcVideoChannelTest, UnsignalledSsrcHasACooldown) {
packet.SetSsrc(kSsrc2);
receive_channel_->OnPacketReceived(packet);
}
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
// The old unsignalled receive stream was destroyed and replaced, so we still
// only have one unsignalled receive stream. But tha packet counter for kSsrc2
@ -7394,7 +7383,7 @@ TEST_F(WebRtcVideoChannelTest,
rtx_packet.SetPayloadType(rtx_vp8_payload_type);
rtx_packet.SetSsrc(rtx_ssrc);
receive_channel_->OnPacketReceived(rtx_packet);
rtc::Thread::Current()->ProcessMessages(0);
time_controller_.AdvanceTime(TimeDelta::Zero());
EXPECT_EQ(1u, fake_call_->GetVideoReceiveStreams().size());
// Send media packet.
@ -9650,7 +9639,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetSources) {
// Send and receive one frame.
SendFrame();
EXPECT_FRAME_WAIT(1, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(1, kVideoWidth, kVideoHeight);
EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty());
EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1));
@ -9664,7 +9653,7 @@ TEST_F(WebRtcVideoChannelBaseTest, GetSources) {
// Send and receive another frame.
SendFrame();
EXPECT_FRAME_WAIT(2, kVideoWidth, kVideoHeight, kTimeout);
EXPECT_FRAME(2, kVideoWidth, kVideoHeight);
EXPECT_THAT(channel_->GetSources(kSsrc - 1), IsEmpty());
EXPECT_THAT(channel_->GetSources(kSsrc), SizeIs(1));
@ -9717,8 +9706,7 @@ TEST_F(WebRtcVideoChannelBaseTest, EncoderSelectorSwitchCodec) {
.WillRepeatedly(Return(webrtc::SdpVideoFormat("VP9")));
channel_->SetEncoderSelector(kSsrc, &encoder_selector);
rtc::Thread::Current()->ProcessMessages(30);
time_controller_.AdvanceTime(kFrameDuration);
ASSERT_TRUE(channel_->GetSendCodec(&codec));
EXPECT_EQ("VP9", codec.name);