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Create a "slow peerconnection unittests" target

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This CL moves all tests that take more than 5 seconds into the new target.

Bug: webrtc:14025
Change-Id: I760d1a270b399b581f41606647740466f6b87e7c
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/261262
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Reviewed-by: Jeremy Leconte <jleconte@google.com>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#36782}
This commit is contained in:
Harald Alvestrand
2022-05-05 13:21:19 +00:00
committed by WebRTC LUCI CQ
parent 8615bf0582
commit f8f7b70050
8 changed files with 1608 additions and 366 deletions
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259
pc/peer_connection_integrationtest.cc
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@ -8,17 +8,24 @@
* be found in the AUTHORS file in the root of the source tree.
*/
// Integration tests for PeerConnection.
// These tests exercise a full stack over a simulated network.
//
// NOTE: If your test takes a while (guideline: more than 5 seconds),
// do NOT add it here, but instead add it to the file
// slow_peer_connection_integrationtest.cc
#include <stdint.h>
#include <algorithm>
#include <memory>
#include <string>
#include <tuple>
#include <type_traits>
#include <utility>
#include <vector>
#include "absl/algorithm/container.h"
#include "absl/strings/string_view.h"
#include "absl/types/optional.h"
#include "api/async_resolver_factory.h"
#include "api/candidate.h"
@ -55,7 +62,6 @@
#include "p2p/base/port.h"
#include "p2p/base/port_allocator.h"
#include "p2p/base/port_interface.h"
#include "p2p/base/stun_server.h"
#include "p2p/base/test_stun_server.h"
#include "p2p/base/test_turn_customizer.h"
#include "p2p/base/test_turn_server.h"
@ -615,65 +621,6 @@ TEST_P(PeerConnectionIntegrationTest, AddAudioToVideoOnlyCall) {
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a call that's transferred to a new caller with a different
// DTLS fingerprint.
TEST_P(PeerConnectionIntegrationTest, CallTransferredForCallee) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Keep the original peer around which will still send packets to the
// receiving client. These SRTP packets will be dropped.
std::unique_ptr<PeerConnectionIntegrationWrapper> original_peer(
SetCallerPcWrapperAndReturnCurrent(
CreatePeerConnectionWrapperWithAlternateKey().release()));
// TODO(deadbeef): Why do we call Close here? That goes against the comment
// directly above.
original_peer->pc()->Close();
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for some additional frames to be transmitted end-to-end.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a call that's transferred to a new callee with a different
// DTLS fingerprint.
TEST_P(PeerConnectionIntegrationTest, CallTransferredForCaller) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Keep the original peer around which will still send packets to the
// receiving client. These SRTP packets will be dropped.
std::unique_ptr<PeerConnectionIntegrationWrapper> original_peer(
SetCalleePcWrapperAndReturnCurrent(
CreatePeerConnectionWrapperWithAlternateKey().release()));
// TODO(deadbeef): Why do we call Close here? That goes against the comment
// directly above.
original_peer->pc()->Close();
ConnectFakeSignaling();
callee()->AddAudioVideoTracks();
caller()->SetOfferAnswerOptions(IceRestartOfferAnswerOptions());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Wait for some additional frames to be transmitted end-to-end.
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
}
// This test sets up a non-bundled call and negotiates bundling at the same
// time as starting an ICE restart. When bundling is in effect in the restart,
// the DTLS-SRTP context should be successfully reset.
@ -1361,31 +1308,6 @@ TEST_P(PeerConnectionIntegrationTest, GetBytesSentStatsWithOldStatsApi) {
EXPECT_GT(caller()->OldGetStatsForTrack(video_track.get())->BytesSent(), 0);
}
// Test that we can get capture start ntp time.
TEST_P(PeerConnectionIntegrationTest, GetCaptureStartNtpTimeWithOldStatsApi) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
caller()->AddAudioTrack();
callee()->AddAudioTrack();
// Do offer/answer, wait for the callee to receive some frames.
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// Get the remote audio track created on the receiver, so they can be used as
// GetStats filters.
auto receivers = callee()->pc()->GetReceivers();
ASSERT_EQ(1u, receivers.size());
auto remote_audio_track = receivers[0]->track();
// Get the audio output level stats. Note that the level is not available
// until an RTCP packet has been received.
EXPECT_TRUE_WAIT(callee()->OldGetStatsForTrack(remote_audio_track.get())
->CaptureStartNtpTime() > 0,
2 * kMaxWaitForFramesMs);
}
// Test that the track ID is associated with all local and remote SSRC stats
// using the old GetStats() and more than 1 audio and more than 1 video track.
// This is a regression test for crbug.com/906988
@ -2009,106 +1931,6 @@ class PeerConnectionIntegrationIceStatesTestWithFakeClock
#if !defined(THREAD_SANITIZER)
// This test provokes TSAN errors. bugs.webrtc.org/11282
// Tests that the PeerConnection goes through all the ICE gathering/connection
// states over the duration of the call. This includes Disconnected and Failed
// states, induced by putting a firewall between the peers and waiting for them
// to time out.
TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock, VerifyIceStates) {
const SocketAddress kStunServerAddress =
SocketAddress("99.99.99.1", cricket::STUN_SERVER_PORT);
StartStunServer(kStunServerAddress);
PeerConnectionInterface::RTCConfiguration config;
PeerConnectionInterface::IceServer ice_stun_server;
ice_stun_server.urls.push_back(
"stun:" + kStunServerAddress.HostAsURIString() + ":" +
kStunServerAddress.PortAsString());
config.servers.push_back(ice_stun_server);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(config, config));
ConnectFakeSignaling();
SetPortAllocatorFlags();
SetUpNetworkInterfaces();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
// Initial state before anything happens.
ASSERT_EQ(PeerConnectionInterface::kIceGatheringNew,
caller()->ice_gathering_state());
ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
caller()->ice_connection_state());
ASSERT_EQ(PeerConnectionInterface::kIceConnectionNew,
caller()->standardized_ice_connection_state());
// Start the call by creating the offer, setting it as the local description,
// then sending it to the peer who will respond with an answer. This happens
// asynchronously so that we can watch the states as it runs in the
// background.
caller()->CreateAndSetAndSignalOffer();
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout,
FakeClock());
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->standardized_ice_connection_state(),
kDefaultTimeout, FakeClock());
// Verify that the observer was notified of the intermediate transitions.
EXPECT_THAT(caller()->ice_connection_state_history(),
ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
PeerConnectionInterface::kIceConnectionConnected,
PeerConnectionInterface::kIceConnectionCompleted));
EXPECT_THAT(caller()->standardized_ice_connection_state_history(),
ElementsAre(PeerConnectionInterface::kIceConnectionChecking,
PeerConnectionInterface::kIceConnectionConnected,
PeerConnectionInterface::kIceConnectionCompleted));
EXPECT_THAT(
caller()->peer_connection_state_history(),
ElementsAre(PeerConnectionInterface::PeerConnectionState::kConnecting,
PeerConnectionInterface::PeerConnectionState::kConnected));
EXPECT_THAT(caller()->ice_gathering_state_history(),
ElementsAre(PeerConnectionInterface::kIceGatheringGathering,
PeerConnectionInterface::kIceGatheringComplete));
// Block connections to/from the caller and wait for ICE to become
// disconnected.
for (const auto& caller_address : CallerAddresses()) {
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
}
RTC_LOG(LS_INFO) << "Firewall rules applied";
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
caller()->ice_connection_state(), kDefaultTimeout,
FakeClock());
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionDisconnected,
caller()->standardized_ice_connection_state(),
kDefaultTimeout, FakeClock());
// Let ICE re-establish by removing the firewall rules.
firewall()->ClearRules();
RTC_LOG(LS_INFO) << "Firewall rules cleared";
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->ice_connection_state(), kDefaultTimeout,
FakeClock());
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionCompleted,
caller()->standardized_ice_connection_state(),
kDefaultTimeout, FakeClock());
// According to RFC7675, if there is no response within 30 seconds then the
// peer should consider the other side to have rejected the connection. This
// is signaled by the state transitioning to "failed".
constexpr int kConsentTimeout = 30000;
for (const auto& caller_address : CallerAddresses()) {
firewall()->AddRule(false, rtc::FP_ANY, rtc::FD_ANY, caller_address);
}
RTC_LOG(LS_INFO) << "Firewall rules applied again";
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
caller()->ice_connection_state(), kConsentTimeout,
FakeClock());
ASSERT_EQ_SIMULATED_WAIT(PeerConnectionInterface::kIceConnectionFailed,
caller()->standardized_ice_connection_state(),
kConsentTimeout, FakeClock());
}
// Tests that if the connection doesn't get set up properly we eventually reach
// the "failed" iceConnectionState.
TEST_P(PeerConnectionIntegrationIceStatesTestWithFakeClock,
@ -2770,71 +2592,6 @@ TEST_P(PeerConnectionIntegrationTest,
EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
}
TEST_P(PeerConnectionIntegrationTest,
SSLCertificateVerifierFailureUsedForTurnConnectionsFailsConnection) {
static const rtc::SocketAddress turn_server_internal_address{"88.88.88.0",
3478};
static const rtc::SocketAddress turn_server_external_address{"88.88.88.1", 0};
// Enable TCP-TLS for the fake turn server. We need to pass in 88.88.88.0 so
// that host name verification passes on the fake certificate.
CreateTurnServer(turn_server_internal_address, turn_server_external_address,
cricket::PROTO_TLS, "88.88.88.0");
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.urls.push_back("turns:88.88.88.0:3478?transport=tcp");
ice_server.username = "test";
ice_server.password = "test";
PeerConnectionInterface::RTCConfiguration client_1_config;
client_1_config.servers.push_back(ice_server);
client_1_config.type = webrtc::PeerConnectionInterface::kRelay;
PeerConnectionInterface::RTCConfiguration client_2_config;
client_2_config.servers.push_back(ice_server);
// Setting the type to kRelay forces the connection to go through a TURN
// server.
client_2_config.type = webrtc::PeerConnectionInterface::kRelay;
// Get a copy to the pointer so we can verify calls later.
rtc::TestCertificateVerifier* client_1_cert_verifier =
new rtc::TestCertificateVerifier();
client_1_cert_verifier->verify_certificate_ = false;
rtc::TestCertificateVerifier* client_2_cert_verifier =
new rtc::TestCertificateVerifier();
client_2_cert_verifier->verify_certificate_ = false;
// Create the dependencies with the test certificate verifier.
webrtc::PeerConnectionDependencies client_1_deps(nullptr);
client_1_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_1_cert_verifier);
webrtc::PeerConnectionDependencies client_2_deps(nullptr);
client_2_deps.tls_cert_verifier =
std::unique_ptr<rtc::TestCertificateVerifier>(client_2_cert_verifier);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfigAndDeps(
client_1_config, std::move(client_1_deps), client_2_config,
std::move(client_2_deps)));
ConnectFakeSignaling();
// Set "offer to receive audio/video" without adding any tracks, so we just
// set up ICE/DTLS with no media.
PeerConnectionInterface::RTCOfferAnswerOptions options;
options.offer_to_receive_audio = 1;
options.offer_to_receive_video = 1;
caller()->SetOfferAnswerOptions(options);
caller()->CreateAndSetAndSignalOffer();
bool wait_res = true;
// TODO(bugs.webrtc.org/9219): When IceConnectionState is implemented
// properly, should be able to just wait for a state of "failed" instead of
// waiting a fixed 10 seconds.
WAIT_(DtlsConnected(), kDefaultTimeout, wait_res);
ASSERT_FALSE(wait_res);
EXPECT_GT(client_1_cert_verifier->call_count_, 0u);
EXPECT_GT(client_2_cert_verifier->call_count_, 0u);
}
// Test that the injected ICE transport factory is used to create ICE transports
// for WebRTC connections.
TEST_P(PeerConnectionIntegrationTest, IceTransportFactoryUsedForConnections) {