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Remove flaky test cases from peerconnection_unittest. The chain of API calls is tested from top to bottom anyway.

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BUG=3871
R=mflodman@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/41879004

Cr-Commit-Position: refs/heads/master@{#8359}
git-svn-id: http://webrtc.googlecode.com/svn/trunk@8359 4adac7df-926f-26a2-2b94-8c16560cd09d
This commit is contained in:
solenberg@webrtc.org 2015-02-13 11:09:20 +00:00
parent ba97ea69f0
commit 40fdb8ab96
2 changed files with 0 additions and 95 deletions
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92
talk/app/webrtc/peerconnection_unittest.cc
View File

@ -92,7 +92,6 @@ static const int kMaxWaitMs = 2000;
// warnings.
#if !defined(THREAD_SANITIZER)
static const int kMaxWaitForStatsMs = 3000;
static const int kMaxWaitForRembMs = 5000;
#endif
static const int kMaxWaitForFramesMs = 10000;
static const int kEndAudioFrameCount = 3;
@ -1038,30 +1037,6 @@ class P2PTestConductor : public testing::Test {
}
}
// Wait until 'size' bytes of audio has been seen by the receiver, on the
// first audio stream.
void WaitForAudioData(int size) {
const int kMaxWaitForAudioDataMs = 10000;
StreamCollectionInterface* local_streams =
initializing_client()->local_streams();
ASSERT_GT(local_streams->count(), 0u);
ASSERT_GT(local_streams->at(0)->GetAudioTracks().size(), 0u);
MediaStreamTrackInterface* local_audio_track =
local_streams->at(0)->GetAudioTracks()[0];
// Wait until *any* audio has been received.
EXPECT_TRUE_WAIT(
receiving_client()->GetBytesReceivedStats(local_audio_track) > 0,
kMaxWaitForAudioDataMs);
// Wait until 'size' number of bytes have been received.
size += receiving_client()->GetBytesReceivedStats(local_audio_track);
EXPECT_TRUE_WAIT(
receiving_client()->GetBytesReceivedStats(local_audio_track) > size,
kMaxWaitForAudioDataMs);
}
SignalingClass* initializing_client() { return initiating_client_.get(); }
SignalingClass* receiving_client() { return receiving_client_.get(); }
@ -1472,7 +1447,6 @@ TEST_F(JsepPeerConnectionP2PTestClient, IceRestart) {
EXPECT_NE(receiver_candidate, receiver_candidate_restart);
}
// This test sets up a Jsep call between two parties with external
// VideoDecoderFactory.
// TODO(holmer): Disabled due to sometimes crashing on buildbots.
@ -1484,70 +1458,4 @@ TEST_F(JsepPeerConnectionP2PTestClient,
LocalP2PTest();
}
// Test receive bandwidth stats with only audio enabled at receiver.
TEST_F(JsepPeerConnectionP2PTestClient, ReceivedBweStatsAudio) {
ASSERT_TRUE(CreateTestClients());
receiving_client()->SetReceiveAudioVideo(true, false);
LocalP2PTest();
// Wait until we have received some audio data. Following REMB shoud be zero.
WaitForAudioData(10000);
EXPECT_EQ_WAIT(
receiving_client()->GetAvailableReceivedBandwidthStats(), 0,
kMaxWaitForRembMs);
}
// Test receive bandwidth stats with combined BWE.
// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3871.
TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_ReceivedBweStatsCombined) {
FakeConstraints setup_constraints;
setup_constraints.AddOptional(
MediaConstraintsInterface::kCombinedAudioVideoBwe, true);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
initializing_client()->AddMediaStream(true, true);
initializing_client()->AddMediaStream(false, true);
initializing_client()->AddMediaStream(false, true);
initializing_client()->AddMediaStream(false, true);
LocalP2PTest();
// Run until a non-zero bw is reported.
EXPECT_TRUE_WAIT(receiving_client()->GetAvailableReceivedBandwidthStats() > 0,
kMaxWaitForRembMs);
// Halt video capturers, then run until we have gotten some audio. Following
// REMB should be non-zero.
initializing_client()->StopVideoCapturers();
WaitForAudioData(10000);
EXPECT_TRUE_WAIT(
receiving_client()->GetAvailableReceivedBandwidthStats() > 0,
kMaxWaitForRembMs);
}
// Test receive bandwidth stats with 1 video, 3 audio streams but no combined
// BWE.
// Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3871.
TEST_F(JsepPeerConnectionP2PTestClient, DISABLED_ReceivedBweStatsNotCombined) {
FakeConstraints setup_constraints;
setup_constraints.AddOptional(
MediaConstraintsInterface::kCombinedAudioVideoBwe, false);
ASSERT_TRUE(CreateTestClients(&setup_constraints, &setup_constraints));
initializing_client()->AddMediaStream(true, true);
initializing_client()->AddMediaStream(false, true);
initializing_client()->AddMediaStream(false, true);
initializing_client()->AddMediaStream(false, true);
LocalP2PTest();
// Run until a non-zero bw is reported.
EXPECT_TRUE_WAIT(receiving_client()->GetAvailableReceivedBandwidthStats() > 0,
kMaxWaitForRembMs);
// Halt video capturers, then run until we have gotten some audio. Following
// REMB should be zero.
initializing_client()->StopVideoCapturers();
WaitForAudioData(10000);
EXPECT_EQ_WAIT(
receiving_client()->GetAvailableReceivedBandwidthStats(), 0,
kMaxWaitForRembMs);
}
#endif // if !defined(THREAD_SANITIZER)

3
tools/valgrind-webrtc/gtest_exclude/libjingle_peerconnection_unittest.gtest-memcheck.txt
View File

@ -20,9 +20,6 @@ JsepPeerConnectionP2PTestClient.LocalP2PTestOfferSdesToDtls
JsepPeerConnectionP2PTestClient.LocalP2PTestOfferDtlsToSdes
JsepPeerConnectionP2PTestClient.LocalP2PTestWithoutMsid
JsepPeerConnectionP2PTestClient.LocalP2PTestWithVideoDecoderFactory
JsepPeerConnectionP2PTestClient.ReceivedBweStatsAudio
JsepPeerConnectionP2PTestClient.ReceivedBweStatsCombined
JsepPeerConnectionP2PTestClient.ReceivedBweStatsNotCombined
JsepPeerConnectionP2PTestClient.RegisterDataChannelObserver
JsepPeerConnectionP2PTestClient.UpdateOfferWithRejectedContent
PeerConnectionEndToEndTest.Call