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/BUILD.gn b/pc/BUILD.gn
index 195d185..329d667 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -206,6 +206,7 @@
"../stats",
"../system_wrappers",
"../system_wrappers:field_trial_api",
+ "../system_wrappers:metrics_api",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -296,7 +297,6 @@
":rtc_pc",
":rtc_pc_base",
"../api:array_view",
- "../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../call:rtp_interfaces",
"../logging:rtc_event_log_api",
@@ -334,7 +334,6 @@
]
deps = [
":pc_test_utils",
- "../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
@@ -494,7 +493,6 @@
":pc_test_utils",
"..:webrtc_common",
"../api:callfactory_api",
- "../api:fakemetricsobserver",
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
"../api/audio_codecs:audio_codecs_api",
diff --git a/pc/peerconnection.cc b/pc/peerconnection.cc
index e205953..a137f05 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -52,6 +52,7 @@
#include "rtc_base/trace_event.h"
#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/field_trial.h"
+#include "system_wrappers/include/metrics.h"
using cricket::ContentInfo;
using cricket::ContentInfos;
@@ -383,15 +384,10 @@
return desc1.contents().size() == desc2.contents().size();
}
-void NoteKeyProtocolAndMedia(
- KeyExchangeProtocolType protocol_type,
- cricket::MediaType media_type,
- rtc::scoped_refptr<webrtc::UMAObserver> uma_observer) {
- if (!uma_observer)
- return;
- uma_observer->IncrementEnumCounter(webrtc::kEnumCounterKeyProtocol,
- protocol_type,
- webrtc::kEnumCounterKeyProtocolMax);
+void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
+ cricket::MediaType media_type) {
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
+ kEnumCounterKeyProtocolMax);
static const std::map<std::pair<KeyExchangeProtocolType, cricket::MediaType>,
KeyExchangeProtocolMedia>
proto_media_counter_map = {
@@ -410,9 +406,8 @@
auto it = proto_media_counter_map.find({protocol_type, media_type});
if (it != proto_media_counter_map.end()) {
- uma_observer->IncrementEnumCounter(webrtc::kEnumCounterKeyProtocolMediaType,
- it->second,
- kEnumCounterKeyProtocolMediaTypeMax);
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
+ it->second, kEnumCounterKeyProtocolMediaTypeMax);
}
}
@@ -422,9 +417,7 @@
// needs a ufrag and pwd. Mismatches, such as replying with a DTLS fingerprint
// to SDES keys, will be caught in JsepTransport negotiation, and backstopped
// by Channel's |srtp_required| check.
-RTCError VerifyCrypto(const SessionDescription* desc,
- bool dtls_enabled,
- rtc::scoped_refptr<webrtc::UMAObserver> uma_observer) {
+RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
for (const cricket::ContentInfo& content_info : desc->contents()) {
@@ -434,8 +427,7 @@
// Note what media is used with each crypto protocol, for all sections.
NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
: webrtc::kEnumCounterKeyProtocolSdes,
- content_info.media_description()->type(),
- uma_observer);
+ content_info.media_description()->type());
const std::string& mid = content_info.name;
if (bundle && bundle->HasContentName(mid) &&
mid != *(bundle->FirstContentName())) {
@@ -939,6 +931,16 @@
NoteUsageEvent(UsageEvent::TURN_SERVER_ADDED);
}
+ // Send information about IPv4/IPv6 status.
+ PeerConnectionAddressFamilyCounter address_family;
+ if (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6) {
+ address_family = kPeerConnection_IPv6;
+ } else {
+ address_family = kPeerConnection_IPv4;
+ }
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics", address_family,
+ kPeerConnectionAddressFamilyCounter_Max);
+
const PeerConnectionFactoryInterface::Options& options = factory_->options();
// RFC 3264: The numeric value of the session id and version in the
@@ -3045,18 +3047,6 @@
network_thread()->Invoke<void>(
RTC_FROM_HERE,
rtc::Bind(&PeerConnection::SetMetricObserver_n, this, observer));
- // Send information about IPv4/IPv6 status.
- if (uma_observer_) {
- if (port_allocator_flags_ & cricket::PORTALLOCATOR_ENABLE_IPV6) {
- uma_observer_->IncrementEnumCounter(
- kEnumCounterAddressFamily, kPeerConnection_IPv6,
- kPeerConnectionAddressFamilyCounter_Max);
- } else {
- uma_observer_->IncrementEnumCounter(
- kEnumCounterAddressFamily, kPeerConnection_IPv4,
- kPeerConnectionAddressFamilyCounter_Max);
- }
- }
}
void PeerConnection::SetMetricObserver_n(UMAObserver* observer) {
@@ -5284,9 +5274,7 @@
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
- if (metrics_observer()) {
- ReportTransportStats();
- }
+ ReportTransportStats();
break;
default:
RTC_NOTREACHED();
@@ -5338,11 +5326,9 @@
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
- if (metrics_observer()) {
- metrics_observer()->IncrementEnumCounter(
- webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
- static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
- }
+ RTC_HISTOGRAM_ENUMERATION(
+ "WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
+ static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
}
void PeerConnection::EnableSending() {
@@ -5780,8 +5766,7 @@
std::string crypto_error;
if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
dtls_enabled_) {
- RTCError crypto_error =
- VerifyCrypto(sdesc->description(), dtls_enabled_, uma_observer_);
+ RTCError crypto_error = VerifyCrypto(sdesc->description(), dtls_enabled_);
if (!crypto_error.ok()) {
return crypto_error;
}
@@ -5908,9 +5893,6 @@
void PeerConnection::ReportSdpFormatReceived(
const SessionDescriptionInterface& remote_offer) {
- if (!uma_observer_) {
- return;
- }
int num_audio_mlines = 0;
int num_video_mlines = 0;
int num_audio_tracks = 0;
@@ -5935,8 +5917,8 @@
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
format = kSdpFormatReceivedSimple;
}
- uma_observer_->IncrementEnumCounter(kEnumCounterSdpFormatReceived, format,
- kSdpFormatReceivedMax);
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
+ kSdpFormatReceivedMax);
}
void PeerConnection::NoteUsageEvent(UsageEvent event) {
@@ -5946,42 +5928,33 @@
void PeerConnection::ReportUsagePattern() const {
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
- if (uma_observer_) {
- uma_observer_->IncrementSparseEnumCounter(kEnumCounterUsagePattern,
- usage_event_accumulator_);
- }
+ RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
+ usage_event_accumulator_,
+ static_cast<int>(UsageEvent::MAX_VALUE));
}
void PeerConnection::ReportNegotiatedSdpSemantics(
const SessionDescriptionInterface& answer) {
- if (!uma_observer_) {
- return;
- }
+ SdpSemanticNegotiated semantics_negotiated;
switch (answer.description()->msid_signaling()) {
case 0:
- uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
- kSdpSemanticNegotiatedNone,
- kSdpSemanticNegotiatedMax);
+ semantics_negotiated = kSdpSemanticNegotiatedNone;
break;
case cricket::kMsidSignalingMediaSection:
- uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
- kSdpSemanticNegotiatedUnifiedPlan,
- kSdpSemanticNegotiatedMax);
+ semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
break;
case cricket::kMsidSignalingSsrcAttribute:
- uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
- kSdpSemanticNegotiatedPlanB,
- kSdpSemanticNegotiatedMax);
+ semantics_negotiated = kSdpSemanticNegotiatedPlanB;
break;
case cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute:
- uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
- kSdpSemanticNegotiatedMixed,
- kSdpSemanticNegotiatedMax);
+ semantics_negotiated = kSdpSemanticNegotiatedMixed;
break;
default:
RTC_NOTREACHED();
}
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpSemanticNegotiated",
+ semantics_negotiated, kSdpSemanticNegotiatedMax);
}
// We need to check the local/remote description for the Transport instead of
@@ -6075,7 +6048,6 @@
// for IPv4 and IPv6.
void PeerConnection::ReportBestConnectionState(
const cricket::TransportStats& stats) {
- RTC_DCHECK(metrics_observer());
for (const cricket::TransportChannelStats& channel_stats :
stats.channel_stats) {
for (const cricket::ConnectionInfo& connection_info :
@@ -6084,7 +6056,6 @@
continue;
}
- PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
@@ -6092,26 +6063,26 @@
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.type() == RELAY_PORT_TYPE &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
- type = kEnumCounterIceCandidatePairTypeTcp;
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
+ GetIceCandidatePairCounter(local, remote),
+ kIceCandidatePairMax);
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
- type = kEnumCounterIceCandidatePairTypeUdp;
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
+ GetIceCandidatePairCounter(local, remote),
+ kIceCandidatePairMax);
} else {
RTC_CHECK(0);
}
- metrics_observer()->IncrementEnumCounter(
- type, GetIceCandidatePairCounter(local, remote),
- kIceCandidatePairMax);
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
- metrics_observer()->IncrementEnumCounter(
- kEnumCounterAddressFamily, kBestConnections_IPv4,
- kPeerConnectionAddressFamilyCounter_Max);
-
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
+ kBestConnections_IPv4,
+ kPeerConnectionAddressFamilyCounter_Max);
} else if (local.address().family() == AF_INET6) {
- metrics_observer()->IncrementEnumCounter(
- kEnumCounterAddressFamily, kBestConnections_IPv6,
- kPeerConnectionAddressFamilyCounter_Max);
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
+ kBestConnections_IPv6,
+ kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(0);
}
@@ -6124,7 +6095,6 @@
void PeerConnection::ReportNegotiatedCiphers(
const cricket::TransportStats& stats,
const std::set<cricket::MediaType>& media_types) {
- RTC_DCHECK(metrics_observer());
if (!dtls_enabled_ || stats.channel_stats.empty()) {
return;
}
@@ -6136,33 +6106,53 @@
return;
}
- for (cricket::MediaType media_type : media_types) {
- PeerConnectionEnumCounterType srtp_counter_type;
- PeerConnectionEnumCounterType ssl_counter_type;
- switch (media_type) {
- case cricket::MEDIA_TYPE_AUDIO:
- srtp_counter_type = kEnumCounterAudioSrtpCipher;
- ssl_counter_type = kEnumCounterAudioSslCipher;
- break;
- case cricket::MEDIA_TYPE_VIDEO:
- srtp_counter_type = kEnumCounterVideoSrtpCipher;
- ssl_counter_type = kEnumCounterVideoSslCipher;
- break;
- case cricket::MEDIA_TYPE_DATA:
- srtp_counter_type = kEnumCounterDataSrtpCipher;
- ssl_counter_type = kEnumCounterDataSslCipher;
- break;
- default:
- RTC_NOTREACHED();
- continue;
+ if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
+ for (cricket::MediaType media_type : media_types) {
+ switch (media_type) {
+ case cricket::MEDIA_TYPE_AUDIO:
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
+ "WebRTC.PeerConnection.SrtpCryptoSuite.Audio", srtp_crypto_suite,
+ rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
+ break;
+ case cricket::MEDIA_TYPE_VIDEO:
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
+ "WebRTC.PeerConnection.SrtpCryptoSuite.Video", srtp_crypto_suite,
+ rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
+ break;
+ case cricket::MEDIA_TYPE_DATA:
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
+ "WebRTC.PeerConnection.SrtpCryptoSuite.Data", srtp_crypto_suite,
+ rtc::SRTP_CRYPTO_SUITE_MAX_VALUE);
+ break;
+ default:
+ RTC_NOTREACHED();
+ continue;
+ }
}
- if (srtp_crypto_suite != rtc::SRTP_INVALID_CRYPTO_SUITE) {
- metrics_observer()->IncrementSparseEnumCounter(srtp_counter_type,
- srtp_crypto_suite);
- }
- if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
- metrics_observer()->IncrementSparseEnumCounter(ssl_counter_type,
- ssl_cipher_suite);
+ }
+
+ if (ssl_cipher_suite != rtc::TLS_NULL_WITH_NULL_NULL) {
+ for (cricket::MediaType media_type : media_types) {
+ switch (media_type) {
+ case cricket::MEDIA_TYPE_AUDIO:
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
+ "WebRTC.PeerConnection.SslCipherSuite.Audio", ssl_cipher_suite,
+ rtc::SSL_CIPHER_SUITE_MAX_VALUE);
+ break;
+ case cricket::MEDIA_TYPE_VIDEO:
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
+ "WebRTC.PeerConnection.SslCipherSuite.Video", ssl_cipher_suite,
+ rtc::SSL_CIPHER_SUITE_MAX_VALUE);
+ break;
+ case cricket::MEDIA_TYPE_DATA:
+ RTC_HISTOGRAM_ENUMERATION_SPARSE(
+ "WebRTC.PeerConnection.SslCipherSuite.Data", ssl_cipher_suite,
+ rtc::SSL_CIPHER_SUITE_MAX_VALUE);
+ break;
+ default:
+ RTC_NOTREACHED();
+ continue;
+ }
}
}
}
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 1cd9c6c..33abbc7 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -66,7 +66,8 @@
CANDIDATE_COLLECTED = 0x80,
REMOTE_CANDIDATE_ADDED = 0x100,
ICE_STATE_CONNECTED = 0x200,
- CLOSE_CALLED = 0x400
+ CLOSE_CALLED = 0x400,
+ MAX_VALUE = 0x800,
};
explicit PeerConnection(PeerConnectionFactory* factory,
diff --git a/pc/peerconnection_histogram_unittest.cc b/pc/peerconnection_histogram_unittest.cc
index beb989e..049d4c5 100644
--- a/pc/peerconnection_histogram_unittest.cc
+++ b/pc/peerconnection_histogram_unittest.cc
@@ -11,7 +11,6 @@
#include <tuple>
#include "absl/memory/memory.h"
-#include "api/fakemetricsobserver.h"
#include "api/jsep.h"
#include "api/peerconnectionproxy.h"
#include "media/base/fakemediaengine.h"
@@ -23,6 +22,7 @@
#include "pc/test/fakesctptransport.h"
#include "rtc_base/gunit.h"
#include "rtc_base/virtualsocketserver.h"
+#include "system_wrappers/include/metrics_default.h"
namespace webrtc {
@@ -160,6 +160,7 @@
PeerConnectionUsageHistogramTest()
: vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
+ webrtc::metrics::Reset();
}
WrapperPtr CreatePeerConnection() {
@@ -208,14 +209,12 @@
TEST_F(PeerConnectionUsageHistogramTest, UsageFingerprintHistogramFromTimeout) {
auto pc = CreatePeerConnectionWithImmediateReport();
- // Register UMA observer before signaling begins.
- rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
- new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
- pc->GetInternalPeerConnection()->RegisterUMAObserver(caller_observer);
int expected_fingerprint = MakeUsageFingerprint({});
- ASSERT_TRUE_WAIT(caller_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint),
- kDefaultTimeout);
+ ASSERT_TRUE_WAIT(
+ 1u == webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"),
+ kDefaultTimeout);
+ EXPECT_EQ(1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
+ expected_fingerprint));
}
#ifndef WEBRTC_ANDROID
@@ -226,9 +225,6 @@
TEST_F(PeerConnectionUsageHistogramTest, FingerprintAudioVideo) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- // Register UMA observer before signaling begins.
- auto caller_observer = caller->RegisterFakeMetricsObserver();
- auto callee_observer = callee->RegisterFakeMetricsObserver();
caller->AddAudioTrack("audio");
caller->AddVideoTrack("video");
ASSERT_TRUE(caller->ConnectTo(callee.get()));
@@ -243,19 +239,16 @@
PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint));
- EXPECT_TRUE(callee_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint));
+ EXPECT_EQ(2,
+ webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
+ EXPECT_EQ(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
+ expected_fingerprint));
}
#ifdef HAVE_SCTP
TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
auto caller = CreatePeerConnection();
auto callee = CreatePeerConnection();
- // Register UMA observer before signaling begins.
- auto caller_observer = caller->RegisterFakeMetricsObserver();
- auto callee_observer = callee->RegisterFakeMetricsObserver();
caller->CreateDataChannel("foodata");
ASSERT_TRUE(caller->ConnectTo(callee.get()));
ASSERT_TRUE_WAIT(callee->HaveDataChannel(), kDefaultTimeout);
@@ -269,10 +262,10 @@
PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint));
- EXPECT_TRUE(callee_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint));
+ EXPECT_EQ(2,
+ webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
+ EXPECT_EQ(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
+ expected_fingerprint));
}
#endif // HAVE_SCTP
#endif // WEBRTC_ANDROID
@@ -288,14 +281,15 @@
configuration.servers.push_back(server);
auto caller = CreatePeerConnection(configuration);
ASSERT_TRUE(caller);
- auto caller_observer = caller->RegisterFakeMetricsObserver();
caller->pc()->Close();
int expected_fingerprint =
MakeUsageFingerprint({PeerConnection::UsageEvent::STUN_SERVER_ADDED,
PeerConnection::UsageEvent::TURN_SERVER_ADDED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint));
+ EXPECT_EQ(1,
+ webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
+ EXPECT_EQ(1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
+ expected_fingerprint));
}
TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
@@ -309,7 +303,6 @@
configuration.servers.push_back(server);
auto caller = CreatePeerConnection();
ASSERT_TRUE(caller);
- auto caller_observer = caller->RegisterFakeMetricsObserver();
RTCError error;
caller->pc()->SetConfiguration(configuration, &error);
ASSERT_TRUE(error.ok());
@@ -318,8 +311,10 @@
MakeUsageFingerprint({PeerConnection::UsageEvent::STUN_SERVER_ADDED,
PeerConnection::UsageEvent::TURN_SERVER_ADDED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterUsagePattern, expected_fingerprint));
+ EXPECT_EQ(1,
+ webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
+ EXPECT_EQ(1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
+ expected_fingerprint));
}
} // namespace webrtc
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 |
diff --git a/pc/peerconnection_rtp_unittest.cc b/pc/peerconnection_rtp_unittest.cc
index 50fe0dc..df1b1ee 100644
--- a/pc/peerconnection_rtp_unittest.cc
+++ b/pc/peerconnection_rtp_unittest.cc
@@ -17,7 +17,6 @@
#include "api/jsep.h"
#include "api/mediastreaminterface.h"
#include "api/peerconnectioninterface.h"
-#include "api/umametrics.h"
#include "api/video_codecs/builtin_video_decoder_factory.h"
#include "api/video_codecs/builtin_video_encoder_factory.h"
#include "pc/mediasession.h"
@@ -32,6 +31,7 @@
#include "rtc_base/refcountedobject.h"
#include "rtc_base/scoped_ref_ptr.h"
#include "rtc_base/thread.h"
+#include "system_wrappers/include/metrics_default.h"
#include "test/gmock.h"
// This file contains tests for RTP Media API-related behavior of
@@ -77,7 +77,9 @@
CreateBuiltinVideoEncoderFactory(),
CreateBuiltinVideoDecoderFactory(),
nullptr /* audio_mixer */,
- nullptr /* audio_processing */)) {}
+ nullptr /* audio_processing */)) {
+ webrtc::metrics::Reset();
+ }
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
@@ -1369,7 +1371,6 @@
caller->AddAudioTrack("caller_audio");
auto callee = CreatePeerConnectionWithUnifiedPlan();
callee->AddAudioTrack("callee_audio");
- auto caller_observer = caller->RegisterFakeMetricsObserver();
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -1384,8 +1385,11 @@
EXPECT_EQ(cricket::kMsidSignalingMediaSection,
answer->description()->msid_signaling());
// Check that this is counted correctly
- EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
- kEnumCounterSdpSemanticNegotiated, kSdpSemanticNegotiatedUnifiedPlan));
+ EXPECT_EQ(2, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SdpSemanticNegotiated"));
+ EXPECT_EQ(2, webrtc::metrics::NumEvents(
+ "WebRTC.PeerConnection.SdpSemanticNegotiated",
+ kSdpSemanticNegotiatedUnifiedPlan));
}
TEST_F(PeerConnectionMsidSignalingTest, PlanBOfferToUnifiedPlanAnswer) {
@@ -1470,12 +1474,14 @@
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->CreateDataChannel("dc");
auto callee = CreatePeerConnectionWithUnifiedPlan();
- auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
-
- EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
- kEnumCounterSdpFormatReceived, kSdpFormatReceivedNoTracks));
+ // Note that only the callee does ReportSdpFormatReceived.
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SdpFormatReceived"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
+ kSdpFormatReceivedNoTracks));
}
#endif // HAVE_SCTP
@@ -1484,24 +1490,28 @@
caller->AddAudioTrack("audio");
caller->AddVideoTrack("video");
auto callee = CreatePeerConnectionWithPlanB();
- auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
-
- EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
- kEnumCounterSdpFormatReceived, kSdpFormatReceivedSimple));
+ // Note that only the callee does ReportSdpFormatReceived.
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SdpFormatReceived"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
+ kSdpFormatReceivedSimple));
}
TEST_F(SdpFormatReceivedTest, SimplePlanBIsReportedAsSimple) {
auto caller = CreatePeerConnectionWithPlanB();
caller->AddVideoTrack("video"); // Video only.
auto callee = CreatePeerConnectionWithUnifiedPlan();
- auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
- EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
- kEnumCounterSdpFormatReceived, kSdpFormatReceivedSimple));
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SdpFormatReceived"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
+ kSdpFormatReceivedSimple));
}
TEST_F(SdpFormatReceivedTest, ComplexUnifiedIsReportedAsComplexUnifiedPlan) {
@@ -1510,12 +1520,14 @@
caller->AddAudioTrack("audio2");
caller->AddVideoTrack("video");
auto callee = CreatePeerConnectionWithPlanB();
- auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
-
- EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
- kEnumCounterSdpFormatReceived, kSdpFormatReceivedComplexUnifiedPlan));
+ // Note that only the callee does ReportSdpFormatReceived.
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SdpFormatReceived"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
+ kSdpFormatReceivedComplexUnifiedPlan));
}
TEST_F(SdpFormatReceivedTest, ComplexPlanBIsReportedAsComplexPlanB) {
@@ -1523,15 +1535,17 @@
caller->AddVideoTrack("video1");
caller->AddVideoTrack("video2");
auto callee = CreatePeerConnectionWithUnifiedPlan();
- auto callee_metrics = callee->RegisterFakeMetricsObserver();
// This fails since Unified Plan cannot set a session description with
// multiple "Plan B tracks" in the same media section. But we still expect the
// SDP Format to be recorded.
ASSERT_FALSE(callee->SetRemoteDescription(caller->CreateOffer()));
-
- EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
- kEnumCounterSdpFormatReceived, kSdpFormatReceivedComplexPlanB));
+ // Note that only the callee does ReportSdpFormatReceived.
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SdpFormatReceived"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
+ kSdpFormatReceivedComplexPlanB));
}
// Sender setups in a call.
diff --git a/pc/peerconnectionwrapper.cc b/pc/peerconnectionwrapper.cc
index 0d19cf3..5006337 100644
--- a/pc/peerconnectionwrapper.cc
+++ b/pc/peerconnectionwrapper.cc
@@ -320,12 +320,4 @@
return callback->report();
}
-rtc::scoped_refptr<FakeMetricsObserver>
-PeerConnectionWrapper::RegisterFakeMetricsObserver() {
- RTC_DCHECK(!fake_metrics_observer_);
- fake_metrics_observer_ = new rtc::RefCountedObject<FakeMetricsObserver>();
- pc_->RegisterUMAObserver(fake_metrics_observer_);
- return fake_metrics_observer_;
-}
-
} // namespace webrtc
diff --git a/pc/peerconnectionwrapper.h b/pc/peerconnectionwrapper.h
index f7de67e..436460e8 100644
--- a/pc/peerconnectionwrapper.h
+++ b/pc/peerconnectionwrapper.h
@@ -16,7 +16,6 @@
#include <string>
#include <vector>
-#include "api/fakemetricsobserver.h"
#include "api/peerconnectioninterface.h"
#include "pc/test/mockpeerconnectionobservers.h"
#include "rtc_base/function_view.h"
@@ -171,10 +170,6 @@
// report. If GetStats() fails, this method returns null and fails the test.
rtc::scoped_refptr<const RTCStatsReport> GetStats();
- // Creates a new FakeMetricsObserver and registers it with the PeerConnection
- // as the UMA observer.
- rtc::scoped_refptr<FakeMetricsObserver> RegisterFakeMetricsObserver();
-
private:
std::unique_ptr<SessionDescriptionInterface> CreateSdp(
rtc::FunctionView<void(CreateSessionDescriptionObserver*)> fn,
@@ -185,7 +180,6 @@
rtc::scoped_refptr<PeerConnectionFactoryInterface> pc_factory_;
std::unique_ptr<MockPeerConnectionObserver> observer_;
rtc::scoped_refptr<PeerConnectionInterface> pc_;
- rtc::scoped_refptr<FakeMetricsObserver> fake_metrics_observer_;
};
} // namespace webrtc
diff --git a/pc/srtpsession.cc b/pc/srtpsession.cc
index 3b33f85..28349ad 100644
--- a/pc/srtpsession.cc
+++ b/pc/srtpsession.cc
@@ -139,11 +139,7 @@
int err = srtp_unprotect(session_, p, out_len);
if (err != srtp_err_status_ok) {
RTC_LOG(LS_WARNING) << "Failed to unprotect SRTP packet, err=" << err;
- if (metrics_observer_) {
- metrics_observer_->IncrementSparseEnumCounter(
- webrtc::kEnumCounterSrtpUnprotectError, err);
- }
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.UnprotectSrtpError",
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError",
static_cast<int>(err), kSrtpErrorCodeBoundary);
return false;
}
@@ -161,11 +157,7 @@
int err = srtp_unprotect_rtcp(session_, p, out_len);
if (err != srtp_err_status_ok) {
RTC_LOG(LS_WARNING) << "Failed to unprotect SRTCP packet, err=" << err;
- if (metrics_observer_) {
- metrics_observer_->IncrementSparseEnumCounter(
- webrtc::kEnumCounterSrtcpUnprotectError, err);
- }
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.UnprotectSrtcpError",
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtcpUnprotectError",
static_cast<int>(err), kSrtpErrorCodeBoundary);
return false;
}
diff --git a/pc/srtpsession_unittest.cc b/pc/srtpsession_unittest.cc
index 66e1cea..b1bc9f0 100644
--- a/pc/srtpsession_unittest.cc
+++ b/pc/srtpsession_unittest.cc
@@ -13,20 +13,21 @@
#include <string>
#include "absl/memory/memory.h"
-#include "api/fakemetricsobserver.h"
#include "media/base/fakertp.h"
#include "pc/srtptestutil.h"
#include "rtc_base/gunit.h"
#include "rtc_base/sslstreamadapter.h" // For rtc::SRTP_*
+#include "system_wrappers/include/metrics_default.h"
#include "third_party/libsrtp/include/srtp.h"
namespace rtc {
-using webrtc::FakeMetricsObserver;
-
std::vector<int> kEncryptedHeaderExtensionIds;
class SrtpSessionTest : public testing::Test {
+ public:
+ SrtpSessionTest() { webrtc::metrics::Reset(); }
+
protected:
virtual void SetUp() {
rtp_len_ = sizeof(kPcmuFrame);
@@ -136,9 +137,6 @@
// Test that we fail to unprotect if someone tampers with the RTP/RTCP paylaods.
TEST_F(SrtpSessionTest, TestTamperReject) {
- rtc::scoped_refptr<FakeMetricsObserver> metrics_observer(
- new rtc::RefCountedObject<FakeMetricsObserver>());
- s2_.SetMetricsObserver(metrics_observer);
int out_len;
EXPECT_TRUE(s1_.SetSend(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
kEncryptedHeaderExtensionIds));
@@ -149,29 +147,38 @@
rtp_packet_[0] = 0x12;
rtcp_packet_[1] = 0x34;
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
- EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterSrtpUnprotectError, srtp_err_status_bad_param));
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SrtpUnprotectError"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
+ srtp_err_status_bad_param));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
- EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterSrtcpUnprotectError, srtp_err_status_auth_fail));
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SrtcpUnprotectError"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
+ srtp_err_status_auth_fail));
}
// Test that we fail to unprotect if the payloads are not authenticated.
TEST_F(SrtpSessionTest, TestUnencryptReject) {
- rtc::scoped_refptr<FakeMetricsObserver> metrics_observer(
- new rtc::RefCountedObject<FakeMetricsObserver>());
- s2_.SetMetricsObserver(metrics_observer);
int out_len;
EXPECT_TRUE(s1_.SetSend(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
kEncryptedHeaderExtensionIds));
EXPECT_TRUE(s2_.SetRecv(SRTP_AES128_CM_SHA1_80, kTestKey1, kTestKeyLen,
kEncryptedHeaderExtensionIds));
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
- EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterSrtpUnprotectError, srtp_err_status_auth_fail));
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SrtpUnprotectError"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
+ srtp_err_status_auth_fail));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
- EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
- webrtc::kEnumCounterSrtcpUnprotectError, srtp_err_status_cant_check));
+ EXPECT_EQ(1, webrtc::metrics::NumSamples(
+ "WebRTC.PeerConnection.SrtcpUnprotectError"));
+ EXPECT_EQ(
+ 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
+ srtp_err_status_cant_check));
}
// Test that we fail when using buffers that are too small.
