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Add directive to make webrtc metrics optional.

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Bug: webrtc:11144
Change-Id: I4e75e6aec033784685de3670e880bb9f2b6ee8d2
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/161043
Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Ying Wang <yinwa@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#30040}
This commit is contained in:
Ying Wang
2019-12-09 13:06:53 +01:00
committed by Commit Bot
parent 00d0f178c2
commit ef3998ffd1
26 changed files with 1101 additions and 763 deletions
Download Patch File Download Diff File Expand all files Collapse all files

64
pc/peer_connection_integrationtest.cc
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@ -1658,9 +1658,9 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
expected_cipher_suite));
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
expected_cipher_suite));
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
@ -1844,10 +1844,12 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_METRIC_LE(
2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_METRIC_EQ(
0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
}
// Uses SDES instead of DTLS for key agreement.
@ -1866,10 +1868,12 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_METRIC_LE(
2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_METRIC_EQ(
0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
}
// Basic end-to-end test specifying the |enable_encrypted_rtp_header_extensions|
@ -3036,9 +3040,9 @@ TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
}
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
@ -3058,9 +3062,9 @@ TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
@ -4359,9 +4363,9 @@ TEST_P(PeerConnectionIntegrationTest,
EXPECT_EQ_WAIT(webrtc::PeerConnectionInterface::kIceConnectionConnected,
callee()->ice_connection_state(), kDefaultTimeout);
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostNameHostName));
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostNameHostName));
}
// Test that firewalling the ICE connection causes the clients to identify the
@ -4593,19 +4597,19 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
if (TestIPv6()) {
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
// connection.
EXPECT_EQ(0, num_best_ipv4);
EXPECT_EQ(1, num_best_ipv6);
EXPECT_METRIC_EQ(0, num_best_ipv4);
EXPECT_METRIC_EQ(1, num_best_ipv6);
} else {
EXPECT_EQ(1, num_best_ipv4);
EXPECT_EQ(0, num_best_ipv6);
EXPECT_METRIC_EQ(1, num_best_ipv4);
EXPECT_METRIC_EQ(0, num_best_ipv6);
}
EXPECT_EQ(0, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostHost));
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostPublicHostPublic));
EXPECT_METRIC_EQ(0, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostHost));
EXPECT_METRIC_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostPublicHostPublic));
}
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |