Use GlobalSimulatedTimeController in more webrtc video engine unittests
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}
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index cefc0a9..f46a73a 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -103,6 +103,8 @@
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 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 @@
} // 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 @@
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 @@
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 @@
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 @@
}
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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());
- 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;
- }
- }
+ ASSERT_EQ(2U, send_info.senders.size());
+
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
// 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 @@
// 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 @@
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 @@
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 @@
// 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 @@
// 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 @@
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 @@
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 @@
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 @@
// 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 @@
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 @@
// 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 @@
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 @@
// 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 @@
// 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 @@
// 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 @@
packet.SetSsrc(kSsrc1);
receive_channel_->OnPacketReceived(packet);
}
- rtc::Thread::Current()->ProcessMessages(0);
+
time_controller_.AdvanceTime(
webrtc::TimeDelta::Millis(kUnsignalledReceiveStreamCooldownMs - 1));
@@ -7193,7 +7182,7 @@
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 @@
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 @@
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 @@
// 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 @@
// 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 @@
.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);
