zqz/platform-external-webrtc
1
0
Fork 0
Code Issues Pull Requests Actions Packages Projects Releases Wiki Activity

Revert "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."

Browse Source 
This reverts commit 870bca1f418a1abf445169a638a61f9a649d557f.

Reason for revert: it breaks internal tests and builds

Original change's description:
> Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.
>
> We now use RTC_HISTOGRAM_* macros in system_wrappers/include/metrics.h
> to report the metrics in pc/ and p2p/ that are currently been reported
> using MetricsObserverInterface.
>
> TBR=tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I47c9975402293c72250203fa1ec19eb1668766f6
> Reviewed-on: https://webrtc-review.googlesource.com/83782
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Reviewed-by: Harald Alvestrand <hta@webrtc.org>
> Reviewed-by: Taylor (left Google) <deadbeef@webrtc.org>
> Reviewed-by: Steve Anton <steveanton@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#23914}

TBR=steveanton@webrtc.org,deadbeef@webrtc.org,hta@webrtc.org,tommi@webrtc.org

Change-Id: I1afd92d44f3b8cf3ae9aa6e6daa9a3a272e8097f
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88040
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23916}
This commit is contained in:
Qingsi Wang
2018-07-10 20:53:10 +00:00
committed by Commit Bot
parent 79abc3d61a
commit 13f4c896d5
25 changed files with 510 additions and 307 deletions
Download Patch File Download Diff File Expand all files Collapse all files

94
pc/peerconnection_integrationtest.cc
View File

@ -24,6 +24,7 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/fakemetricsobserver.h"
#include "api/mediastreaminterface.h"
#include "api/peerconnectioninterface.h"
#include "api/peerconnectionproxy.h"
@ -62,7 +63,6 @@
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/testcertificateverifier.h"
#include "rtc_base/virtualsocketserver.h"
#include "system_wrappers/include/metrics_default.h"
#include "test/gmock.h"
using cricket::ContentInfo;
@ -1106,7 +1106,6 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
webrtc::metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
@ -1514,17 +1513,20 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
int expected_cipher_suite) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
callee_options));
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
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_EQ(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
expected_cipher_suite));
caller()->pc()->RegisterUMAObserver(nullptr);
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
@ -1694,6 +1696,9 @@ TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
// Do normal offer/answer and wait for some frames to be received in each
// direction.
@ -1704,10 +1709,12 @@ 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_LE(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_EQ(
0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolSdes));
}
// Uses SDES instead of DTLS for key agreement.
@ -1716,6 +1723,9 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
sdes_config.enable_dtls_srtp.emplace(false);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
ConnectFakeSignaling();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
// Do normal offer/answer and wait for some frames to be received in each
// direction.
@ -1726,10 +1736,12 @@ 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_LE(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_EQ(
0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolDtls));
}
// Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
@ -2731,19 +2743,22 @@ TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
dtls_10_options));
ConnectFakeSignaling();
// Register UMA observer before signaling begins.
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
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_EQ(1,
caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
@ -2753,19 +2768,22 @@ TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
dtls_12_options));
ConnectFakeSignaling();
// Register UMA observer before signaling begins.
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
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_EQ(1,
caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
@ -3484,15 +3502,19 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
SetUpNetworkInterfaces();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer(
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>());
caller()->pc()->RegisterUMAObserver(metrics_observer.get());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
const int num_best_ipv4 = webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
const int num_best_ipv6 = webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
const int num_best_ipv4 = metrics_observer->GetEnumCounter(
webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv4);
const int num_best_ipv6 = metrics_observer->GetEnumCounter(
webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv6);
if (TestIPv6()) {
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
// connection.
@ -3503,12 +3525,12 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
EXPECT_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_EQ(0, metrics_observer->GetEnumCounter(
webrtc::kEnumCounterIceCandidatePairTypeUdp,
webrtc::kIceCandidatePairHostHost));
EXPECT_EQ(1, metrics_observer->GetEnumCounter(
webrtc::kEnumCounterIceCandidatePairTypeUdp,
webrtc::kIceCandidatePairHostPublicHostPublic));
}
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |