Reland "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
This is a reland of 1a2cc0acba6a66f89249455d8e5775849b56f755
Original change's description:
> Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*."
>
> This is a reland of 870bca1f418a1abf445169a638a61f9a649d557f
>
> 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,hta@webrtc.org,tommi@webrtc.org
>
> Bug: webrtc:9409
> Change-Id: I37fc95ced60dea25aa9b4f5ad44bdf7174c8bd5c
> Reviewed-on: https://webrtc-review.googlesource.com/88060
> Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
> Commit-Queue: Qingsi Wang <qingsi@google.com>
> Cr-Commit-Position: refs/heads/master@{#23919}
TBR=steveanton@webrtc.org,tommi@webrtc.org
Bug: webrtc:9409
Change-Id: Ib55f0b6c9bcb9d9585924a4dfac5cf643ff4d76b
Reviewed-on: https://webrtc-review.googlesource.com/88343
Commit-Queue: Qingsi Wang <qingsi@google.com>
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23957}diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 311881a..6def198 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -24,7 +24,6 @@
#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"
@@ -63,6 +62,7 @@
#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;
@@ -1106,6 +1106,7 @@
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
+ webrtc::metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
@@ -1513,20 +1514,17 @@
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(SignalingStateStable(), kDefaultTimeout);
+ ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
- EXPECT_EQ(
- 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
- expected_cipher_suite));
- caller()->pc()->RegisterUMAObserver(nullptr);
+ // TODO(bugs.webrtc.org/9456): Fix it.
+ EXPECT_EQ(1, webrtc::metrics::NumEvents(
+ "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
+ expected_cipher_suite));
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
@@ -1696,9 +1694,6 @@
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.
@@ -1709,12 +1704,10 @@
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
- EXPECT_LE(
- 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
- webrtc::kEnumCounterKeyProtocolDtls));
- EXPECT_EQ(
- 0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
- webrtc::kEnumCounterKeyProtocolSdes));
+ EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
+ webrtc::kEnumCounterKeyProtocolDtls));
+ EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
+ webrtc::kEnumCounterKeyProtocolSdes));
}
// Uses SDES instead of DTLS for key agreement.
@@ -1723,9 +1716,6 @@
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.
@@ -1736,12 +1726,10 @@
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
- EXPECT_LE(
- 1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
- webrtc::kEnumCounterKeyProtocolSdes));
- EXPECT_EQ(
- 0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
- webrtc::kEnumCounterKeyProtocolDtls));
+ EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
+ webrtc::kEnumCounterKeyProtocolSdes));
+ EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
+ webrtc::kEnumCounterKeyProtocolDtls));
}
// Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
@@ -2743,22 +2731,19 @@
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(SignalingStateStable(), kDefaultTimeout);
+ ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
- EXPECT_EQ(1,
- caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
- kDefaultSrtpCryptoSuite));
+ // TODO(bugs.webrtc.org/9456): Fix it.
+ EXPECT_EQ(1, webrtc::metrics::NumEvents(
+ "WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
+ kDefaultSrtpCryptoSuite));
}
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
@@ -2768,22 +2753,19 @@
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(SignalingStateStable(), kDefaultTimeout);
+ ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
- EXPECT_EQ(1,
- caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
- kDefaultSrtpCryptoSuite));
+ // TODO(bugs.webrtc.org/9456): Fix it.
+ EXPECT_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
@@ -3502,19 +3484,15 @@
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);
- 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);
+ // 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);
if (TestIPv6()) {
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
// connection.
@@ -3525,12 +3503,12 @@
EXPECT_EQ(0, num_best_ipv6);
}
- EXPECT_EQ(0, metrics_observer->GetEnumCounter(
- webrtc::kEnumCounterIceCandidatePairTypeUdp,
- webrtc::kIceCandidatePairHostHost));
- EXPECT_EQ(1, metrics_observer->GetEnumCounter(
- webrtc::kEnumCounterIceCandidatePairTypeUdp,
- webrtc::kIceCandidatePairHostPublicHostPublic));
+ EXPECT_EQ(0, webrtc::metrics::NumEvents(
+ "WebRTC.PeerConnection.CandidatePairType_UDP",
+ webrtc::kIceCandidatePairHostHost));
+ EXPECT_EQ(1, webrtc::metrics::NumEvents(
+ "WebRTC.PeerConnection.CandidatePairType_UDP",
+ webrtc::kIceCandidatePairHostPublicHostPublic));
}
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |