Update video tests to clear network interface pointer consistently.
This is related to upcoming changes whereby it will be enforced that
calls to SetInterface(<valid ptr>) and SetInterface(nullptr) be matched
up correctly.
Bug: webrtc:11993
Change-Id: Ic022f9487a7ab297adaced8e620e2384e055673b
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/217241
Commit-Queue: Tommi <tommi@webrtc.org>
Reviewed-by: Danil Chapovalov <danilchap@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33903}
diff --git a/media/engine/webrtc_video_engine_unittest.cc b/media/engine/webrtc_video_engine_unittest.cc
index 4270e27..6d29e3d 100644
--- a/media/engine/webrtc_video_engine_unittest.cc
+++ b/media/engine/webrtc_video_engine_unittest.cc
@@ -1417,6 +1417,10 @@
channel_->SetRecvParameters(parameters);
}
+ ~WebRtcVideoChannelEncodedFrameCallbackTest() override {
+ channel_->SetInterface(nullptr);
+ }
+
void DeliverKeyFrame(uint32_t ssrc) {
webrtc::RtpPacket packet;
packet.SetMarker(true);
@@ -1544,7 +1548,7 @@
webrtc::CreateBuiltinVideoDecoderFactory(),
field_trials_) {}
- virtual void SetUp() {
+ void SetUp() override {
// One testcase calls SetUp in a loop, only create call_ once.
if (!call_) {
webrtc::Call::Config call_config(&event_log_);
@@ -1586,6 +1590,7 @@
// Make the second renderer available for use by a new stream.
EXPECT_TRUE(channel_->SetSink(kSsrc + 2, &renderer2_));
}
+
// Setup an additional stream just to send video. Defer add recv stream.
// This is required if you want to test unsignalled recv of video rtp packets.
void SetUpSecondStreamWithNoRecv() {
@@ -1604,7 +1609,17 @@
EXPECT_TRUE(
channel_->SetVideoSend(kSsrc + 2, nullptr, frame_forwarder_2_.get()));
}
- virtual void TearDown() { channel_.reset(); }
+
+ void TearDown() override {
+ channel_->SetInterface(nullptr);
+ channel_.reset();
+ }
+
+ void ResetTest() {
+ TearDown();
+ SetUp();
+ }
+
bool SetDefaultCodec() { return SetOneCodec(DefaultCodec()); }
bool SetOneCodec(const cricket::VideoCodec& codec) {
@@ -1788,7 +1803,7 @@
const int kCustomRecvBufferSize = 123456;
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-IncreasedReceivebuffers/123456/");
- SetUp();
+ ResetTest();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
@@ -1804,7 +1819,7 @@
const int kCustomRecvBufferSize = 123456;
webrtc::test::ScopedFieldTrials field_trial(
"WebRTC-IncreasedReceivebuffers/123456_Dogfood/");
- SetUp();
+ ResetTest();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
@@ -1824,7 +1839,7 @@
field_trial_string += group;
field_trial_string += "/";
webrtc::test::ScopedFieldTrials field_trial(field_trial_string);
- SetUp();
+ ResetTest();
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
EXPECT_TRUE(SetSend(true));
@@ -1835,8 +1850,6 @@
// Test that stats work properly for a 1-1 call.
TEST_F(WebRtcVideoChannelBaseTest, GetStats) {
- SetUp();
-
const int kDurationSec = 3;
const int kFps = 10;
SendReceiveManyAndGetStats(DefaultCodec(), kDurationSec, kFps);
@@ -1893,8 +1906,6 @@
// Test that stats work properly for a conf call with multiple recv streams.
TEST_F(WebRtcVideoChannelBaseTest, GetStatsMultipleRecvStreams) {
- SetUp();
-
cricket::FakeVideoRenderer renderer1, renderer2;
EXPECT_TRUE(SetOneCodec(DefaultCodec()));
cricket::VideoSendParameters parameters;
@@ -2513,6 +2524,16 @@
ASSERT_TRUE(channel_->SetSendParameters(send_parameters_));
}
+ void TearDown() override {
+ channel_->SetInterface(nullptr);
+ channel_ = nullptr;
+ }
+
+ void ResetTest() {
+ TearDown();
+ SetUp();
+ }
+
cricket::VideoCodec GetEngineCodec(const std::string& name) {
for (const cricket::VideoCodec& engine_codec : engine_.send_codecs()) {
if (absl::EqualsIgnoreCase(name, engine_codec.name))
@@ -3165,7 +3186,7 @@
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-RtcpLossNotification/Enabled/");
- SetUp();
+ ResetTest();
TestLossNotificationState(true);
}
@@ -3173,7 +3194,7 @@
RTC_DCHECK(!override_field_trials_);
override_field_trials_ = std::make_unique<webrtc::test::ScopedFieldTrials>(
"WebRTC-RtcpLossNotification/Enabled/");
- SetUp();
+ ResetTest();
AssignDefaultCodec();
VerifyCodecHasDefaultFeedbackParams(default_codec_, true);
@@ -5248,6 +5269,7 @@
channel->SetInterface(network_interface.get());
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
+ channel->SetInterface(nullptr);
// Default value when DSCP is enabled is also DSCP_DEFAULT, until it is set
// through rtp parameters.
@@ -5277,6 +5299,7 @@
EXPECT_TRUE(static_cast<webrtc::Transport*>(channel.get())
->SendRtcp(kData, sizeof(kData)));
EXPECT_EQ(rtc::DSCP_CS1, network_interface->options().dscp);
+ channel->SetInterface(nullptr);
// Verify that setting the option to false resets the
// DiffServCodePoint.
@@ -5287,6 +5310,7 @@
video_bitrate_allocator_factory_.get())));
channel->SetInterface(network_interface.get());
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
+ channel->SetInterface(nullptr);
}
// This test verifies that the RTCP reduced size mode is properly applied to
diff --git a/media/engine/webrtc_voice_engine_unittest.cc b/media/engine/webrtc_voice_engine_unittest.cc
index 3286837..ecf8c07 100644
--- a/media/engine/webrtc_voice_engine_unittest.cc
+++ b/media/engine/webrtc_voice_engine_unittest.cc
@@ -3203,6 +3203,7 @@
channel->SetInterface(&network_interface);
// Default value when DSCP is disabled should be DSCP_DEFAULT.
EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface.dscp());
+ channel->SetInterface(nullptr);
config.enable_dscp = true;
channel.reset(static_cast<cricket::WebRtcVoiceMediaChannel*>(
@@ -3229,6 +3230,7 @@
const uint8_t kData[10] = {0};
EXPECT_TRUE(channel->SendRtcp(kData, sizeof(kData)));
EXPECT_EQ(rtc::DSCP_CS1, network_interface.options().dscp);
+ channel->SetInterface(nullptr);
// Verify that setting the option to false resets the
// DiffServCodePoint.
