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Fixing and re-enabling some flaky PeerConnection tests.

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BUG=webrtc:3362

Review URL: https://codereview.webrtc.org/1512763003

Cr-Commit-Position: refs/heads/master@{#11016}
This commit is contained in:
deadbeef 2015-12-14 18:27:57 -08:00 committed by Commit bot
parent bd292465ee
commit c9be00797e
2 changed files with 62 additions and 33 deletions
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92
talk/app/webrtc/peerconnection_unittest.cc
View File

@ -171,11 +171,6 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
}
~PeerConnectionTestClient() {
while (!fake_video_renderers_.empty()) {
RenderMap::iterator it = fake_video_renderers_.begin();
delete it->second;
fake_video_renderers_.erase(it);
}
}
void Negotiate() { Negotiate(true, true); }
@ -229,8 +224,8 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
const std::string id = media_stream->GetVideoTracks()[i]->id();
ASSERT_TRUE(fake_video_renderers_.find(id) ==
fake_video_renderers_.end());
fake_video_renderers_[id] =
new webrtc::FakeVideoTrackRenderer(media_stream->GetVideoTracks()[i]);
fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
media_stream->GetVideoTracks()[i]));
}
}
void OnRemoveStream(MediaStreamInterface* media_stream) override {}
@ -265,9 +260,9 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
for (auto it = fake_video_renderers_.begin();
it != fake_video_renderers_.end();) {
if (remote_streams->FindVideoTrack(it->first) == nullptr) {
auto to_delete = it++;
delete to_delete->second;
fake_video_renderers_.erase(to_delete);
auto to_remove = it++;
removed_fake_video_renderers_.push_back(std::move(to_remove->second));
fake_video_renderers_.erase(to_remove);
} else {
++it;
}
@ -284,8 +279,8 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
if (fake_video_renderers_.find(id) != fake_video_renderers_.end()) {
continue;
}
fake_video_renderers_[id] = new webrtc::FakeVideoTrackRenderer(
remote_stream->GetVideoTracks()[track_index]);
fake_video_renderers_[id].reset(new webrtc::FakeVideoTrackRenderer(
remote_stream->GetVideoTracks()[track_index]));
}
}
}
@ -452,6 +447,10 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
return number_of_frames <= fake_audio_capture_module_->frames_received();
}
int audio_frames_received() const {
return fake_audio_capture_module_->frames_received();
}
bool VideoFramesReceivedCheck(int number_of_frames) {
if (video_decoder_factory_enabled_) {
const std::vector<FakeWebRtcVideoDecoder*>& decoders
@ -460,9 +459,8 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
return number_of_frames <= 0;
}
for (std::vector<FakeWebRtcVideoDecoder*>::const_iterator
it = decoders.begin(); it != decoders.end(); ++it) {
if (number_of_frames > (*it)->GetNumFramesReceived()) {
for (FakeWebRtcVideoDecoder* decoder : decoders) {
if (number_of_frames > decoder->GetNumFramesReceived()) {
return false;
}
}
@ -472,9 +470,8 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
return number_of_frames <= 0;
}
for (RenderMap::const_iterator it = fake_video_renderers_.begin();
it != fake_video_renderers_.end(); ++it) {
if (number_of_frames > it->second->num_rendered_frames()) {
for (const auto& pair : fake_video_renderers_) {
if (number_of_frames > pair.second->num_rendered_frames()) {
return false;
}
}
@ -482,6 +479,25 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
}
}
int video_frames_received() const {
int total = 0;
if (video_decoder_factory_enabled_) {
const std::vector<FakeWebRtcVideoDecoder*>& decoders =
fake_video_decoder_factory_->decoders();
for (const FakeWebRtcVideoDecoder* decoder : decoders) {
total += decoder->GetNumFramesReceived();
}
} else {
for (const auto& pair : fake_video_renderers_) {
total += pair.second->num_rendered_frames();
}
for (const auto& renderer : removed_fake_video_renderers_) {
total += renderer->num_rendered_frames();
}
}
return total;
}
// Verify the CreateDtmfSender interface
void VerifyDtmf() {
rtc::scoped_ptr<DummyDtmfObserver> observer(new DummyDtmfObserver());
@ -878,11 +894,14 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver,
std::map<int, IceUfragPwdPair> ice_ufrag_pwd_;
bool expect_ice_restart_ = false;
// Needed to keep track of number of frames send.
// Needed to keep track of number of frames sent.
rtc::scoped_refptr<FakeAudioCaptureModule> fake_audio_capture_module_;
// Needed to keep track of number of frames received.
typedef std::map<std::string, webrtc::FakeVideoTrackRenderer*> RenderMap;
RenderMap fake_video_renderers_;
std::map<std::string, rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
fake_video_renderers_;
// Needed to ensure frames aren't received for removed tracks.
std::vector<rtc::scoped_ptr<webrtc::FakeVideoTrackRenderer>>
removed_fake_video_renderers_;
// Needed to keep track of number of frames received when external decoder
// used.
FakeWebRtcVideoDecoderFactory* fake_video_decoder_factory_ = nullptr;
@ -942,13 +961,26 @@ class P2PTestConductor : public testing::Test {
}
void TestUpdateOfferWithRejectedContent() {
// Renegotiate, rejecting the video m-line.
initiating_client_->Negotiate(true, false);
EXPECT_TRUE_WAIT(
FramesNotPending(kEndAudioFrameCount * 2, kEndVideoFrameCount),
kMaxWaitForFramesMs);
// There shouldn't be any more video frame after the new offer is
// negotiated.
EXPECT_FALSE(VideoFramesReceivedCheck(kEndVideoFrameCount + 1));
ASSERT_TRUE_WAIT(SessionActive(), kMaxWaitForActivationMs);
int pc1_audio_received = initiating_client_->audio_frames_received();
int pc1_video_received = initiating_client_->video_frames_received();
int pc2_audio_received = receiving_client_->audio_frames_received();
int pc2_video_received = receiving_client_->video_frames_received();
// Wait for some additional audio frames to be received.
EXPECT_TRUE_WAIT(initiating_client_->AudioFramesReceivedCheck(
pc1_audio_received + kEndAudioFrameCount) &&
receiving_client_->AudioFramesReceivedCheck(
pc2_audio_received + kEndAudioFrameCount),
kMaxWaitForFramesMs);
// During this time, we shouldn't have received any additional video frames
// for the rejected video tracks.
EXPECT_EQ(pc1_video_received, initiating_client_->video_frames_received());
EXPECT_EQ(pc2_video_received, receiving_client_->video_frames_received());
}
void VerifyRenderedSize(int width, int height) {
@ -1292,8 +1324,7 @@ TEST_F(P2PTestConductor, LocalP2PTestAnswerNone) {
// runs for a while (10 frames), the caller sends an update offer with video
// being rejected. Once the re-negotiation is done, the video flow should stop
// and the audio flow should continue.
// Disabled due to b/14955157.
TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) {
TEST_F(P2PTestConductor, UpdateOfferWithRejectedContent) {
ASSERT_TRUE(CreateTestClients());
LocalP2PTest();
TestUpdateOfferWithRejectedContent();
@ -1301,8 +1332,7 @@ TEST_F(P2PTestConductor, DISABLED_UpdateOfferWithRejectedContent) {
// This test sets up a Jsep call between two parties. The MSID is removed from
// the SDP strings from the caller.
// Disabled due to b/14955157.
TEST_F(P2PTestConductor, DISABLED_LocalP2PTestWithoutMsid) {
TEST_F(P2PTestConductor, LocalP2PTestWithoutMsid) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->RemoveMsidFromReceivedSdp(true);
// TODO(perkj): Currently there is a bug that cause audio to stop playing if

3
talk/app/webrtc/peerconnectionendtoend_unittest.cc
View File

@ -230,8 +230,7 @@ TEST_F(PeerConnectionEndToEndTest, Call) {
}
#endif // if !defined(THREAD_SANITIZER) && !defined(WEBRTC_MAC)
// Disabled per b/14899892
TEST_F(PeerConnectionEndToEndTest, DISABLED_CallWithLegacySdp) {
TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) {
FakeConstraints pc_constraints;
pc_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp,
false);