Revert "Reland "Replace the usage of MetricsObserverInterface by RTC_HISTOGRAM_*.""
This reverts commit 1a2cc0acba6a66f89249455d8e5775849b56f755.
Reason for revert: It breaks internal Android debug build. Need further investigation.
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,deadbeef@webrtc.org,tommi@webrtc.org,hta@webrtc.org,qingsi@google.com,qingsi@webrtc.org
Change-Id: I4a75fc7f52bfd0780526537a5a9a016fb9c20d6a
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:9409
Reviewed-on: https://webrtc-review.googlesource.com/88320
Reviewed-by: Qingsi Wang <qingsi@webrtc.org>
Commit-Queue: Qingsi Wang <qingsi@google.com>
Cr-Commit-Position: refs/heads/master@{#23938}diff --git a/api/BUILD.gn b/api/BUILD.gn
index 55d126a..4357947 100644
--- a/api/BUILD.gn
+++ b/api/BUILD.gn
@@ -86,6 +86,7 @@
"statstypes.cc",
"statstypes.h",
"turncustomizer.h",
+ "umametrics.cc",
"umametrics.h",
"videosourceproxy.h",
]
@@ -451,6 +452,26 @@
]
}
+ rtc_source_set("fakemetricsobserver") {
+ testonly = true
+ sources = [
+ "fakemetricsobserver.cc",
+ "fakemetricsobserver.h",
+ ]
+ deps = [
+ "../media:rtc_media_base",
+ "../rtc_base:checks",
+ "../rtc_base:rtc_base_approved",
+ ]
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+ if (!build_with_mozilla) {
+ deps += [ ":libjingle_peerconnection_api" ]
+ }
+ }
+
rtc_source_set("rtc_api_unittests") {
testonly = true
diff --git a/api/fakemetricsobserver.cc b/api/fakemetricsobserver.cc
new file mode 100644
index 0000000..cd8de39
--- /dev/null
+++ b/api/fakemetricsobserver.cc
@@ -0,0 +1,87 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/fakemetricsobserver.h"
+#include "rtc_base/checks.h"
+
+namespace webrtc {
+
+FakeMetricsObserver::FakeMetricsObserver() {
+ Reset();
+}
+
+void FakeMetricsObserver::Reset() {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ counters_.clear();
+ memset(histogram_samples_, 0, sizeof(histogram_samples_));
+}
+
+void FakeMetricsObserver::IncrementEnumCounter(
+ PeerConnectionEnumCounterType type,
+ int counter,
+ int counter_max) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (counters_.size() <= static_cast<size_t>(type)) {
+ counters_.resize(type + 1);
+ }
+ auto& counters = counters_[type];
+ ++counters[counter];
+}
+
+void FakeMetricsObserver::AddHistogramSample(PeerConnectionMetricsName type,
+ int value) {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ RTC_DCHECK_EQ(histogram_samples_[type], 0);
+ histogram_samples_[type] = value;
+}
+
+int FakeMetricsObserver::GetEnumCounter(PeerConnectionEnumCounterType type,
+ int counter) const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (counters_.size() <= static_cast<size_t>(type)) {
+ return 0;
+ }
+ const auto& it = counters_[type].find(counter);
+ if (it == counters_[type].end()) {
+ return 0;
+ }
+ return it->second;
+}
+
+int FakeMetricsObserver::GetHistogramSample(
+ PeerConnectionMetricsName type) const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ return histogram_samples_[type];
+}
+
+bool FakeMetricsObserver::ExpectOnlySingleEnumCount(
+ PeerConnectionEnumCounterType type,
+ int counter) const {
+ RTC_DCHECK(thread_checker_.CalledOnValidThread());
+ if (counters_.size() <= static_cast<size_t>(type)) {
+ // If a counter has not been allocated then there has been no call to
+ // |IncrementEnumCounter| so all the values are 0.
+ return false;
+ }
+ bool pass = true;
+ if (GetEnumCounter(type, counter) != 1) {
+ RTC_LOG(LS_ERROR) << "Expected single count for counter: " << counter;
+ pass = false;
+ }
+ for (const auto& entry : counters_[type]) {
+ if (entry.first != counter && entry.second > 0) {
+ RTC_LOG(LS_ERROR) << "Expected no count for counter: " << entry.first;
+ pass = false;
+ }
+ }
+ return pass;
+}
+
+} // namespace webrtc
diff --git a/api/fakemetricsobserver.h b/api/fakemetricsobserver.h
new file mode 100644
index 0000000..1f5b704
--- /dev/null
+++ b/api/fakemetricsobserver.h
@@ -0,0 +1,56 @@
+/*
+ * Copyright 2015 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#ifndef API_FAKEMETRICSOBSERVER_H_
+#define API_FAKEMETRICSOBSERVER_H_
+
+#include <map>
+#include <string>
+#include <vector>
+
+#include "api/peerconnectioninterface.h"
+#include "rtc_base/thread_checker.h"
+
+namespace webrtc {
+
+class FakeMetricsObserver : public MetricsObserverInterface {
+ public:
+ FakeMetricsObserver();
+ void Reset();
+
+ void IncrementEnumCounter(PeerConnectionEnumCounterType,
+ int counter,
+ int counter_max) override;
+ void AddHistogramSample(PeerConnectionMetricsName type, int value) override;
+
+ // Accessors to be used by the tests.
+ int GetEnumCounter(PeerConnectionEnumCounterType type, int counter) const;
+ int GetHistogramSample(PeerConnectionMetricsName type) const;
+
+ // Returns true if and only if there is a count of 1 for the given counter and
+ // a count of 0 for all other counters of the given enum type.
+ bool ExpectOnlySingleEnumCount(PeerConnectionEnumCounterType type,
+ int counter) const;
+
+ protected:
+ ~FakeMetricsObserver() {}
+
+ private:
+ rtc::ThreadChecker thread_checker_;
+ // The vector contains maps for each counter type. In the map, it's a mapping
+ // from individual counter to its count, such that it's memory efficient when
+ // comes to sparse enum types, like the SSL ciphers in the IANA registry.
+ std::vector<std::map<int, int>> counters_;
+ int histogram_samples_[kPeerConnectionMetricsName_Max];
+};
+
+} // namespace webrtc
+
+#endif // API_FAKEMETRICSOBSERVER_H_
diff --git a/api/umametrics.cc b/api/umametrics.cc
new file mode 100644
index 0000000..d5f2bb6
--- /dev/null
+++ b/api/umametrics.cc
@@ -0,0 +1,21 @@
+/*
+ * Copyright 2017 The WebRTC project authors. All Rights Reserved.
+ *
+ * Use of this source code is governed by a BSD-style license
+ * that can be found in the LICENSE file in the root of the source
+ * tree. An additional intellectual property rights grant can be found
+ * in the file PATENTS. All contributing project authors may
+ * be found in the AUTHORS file in the root of the source tree.
+ */
+
+#include "api/umametrics.h"
+
+namespace webrtc {
+
+void MetricsObserverInterface::IncrementSparseEnumCounter(
+ PeerConnectionEnumCounterType type,
+ int counter) {
+ IncrementEnumCounter(type, counter, 0 /* Ignored */);
+}
+
+} // namespace webrtc
diff --git a/api/umametrics.h b/api/umametrics.h
index b999b64..081b515 100644
--- a/api/umametrics.h
+++ b/api/umametrics.h
@@ -176,13 +176,13 @@
// number after the highest counter.
virtual void IncrementEnumCounter(PeerConnectionEnumCounterType type,
int counter,
- int counter_max) = 0;
+ int counter_max) {}
// This is used to handle sparse counters like SSL cipher suites.
// TODO(guoweis): Remove the implementation once the dependency's interface
// definition is updated.
virtual void IncrementSparseEnumCounter(PeerConnectionEnumCounterType type,
- int counter) = 0;
+ int counter);
virtual void AddHistogramSample(PeerConnectionMetricsName type,
int value) = 0;
diff --git a/p2p/BUILD.gn b/p2p/BUILD.gn
index 4763001..a2f5160 100644
--- a/p2p/BUILD.gn
+++ b/p2p/BUILD.gn
@@ -91,7 +91,6 @@
"../rtc_base:safe_minmax",
"../rtc_base:stringutils",
"../system_wrappers:field_trial_api",
- "../system_wrappers:metrics_api",
"//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/types:optional",
]
@@ -172,13 +171,13 @@
deps = [
":p2p_test_utils",
":rtc_p2p",
+ "../api:fakemetricsobserver",
"../api:ortc_api",
"../rtc_base:checks",
"../rtc_base:rtc_base",
"../rtc_base:rtc_base_approved",
"../rtc_base:rtc_base_tests_utils",
"../rtc_base:stringutils",
- "../system_wrappers:metrics_default",
"../test:test_support",
"//testing/gtest",
"//third_party/abseil-cpp/absl/memory",
diff --git a/p2p/base/p2ptransportchannel.cc b/p2p/base/p2ptransportchannel.cc
index 67bcdfa..dca8d63 100644
--- a/p2p/base/p2ptransportchannel.cc
+++ b/p2p/base/p2ptransportchannel.cc
@@ -28,7 +28,6 @@
#include "rtc_base/stringencode.h"
#include "rtc_base/timeutils.h"
#include "system_wrappers/include/field_trial.h"
-#include "system_wrappers/include/metrics.h"
namespace {
@@ -676,7 +675,7 @@
SignalGatheringState(this);
}
- if (!allocator_sessions_.empty()) {
+ if (metrics_observer_ && !allocator_sessions_.empty()) {
IceRestartState state;
if (writable()) {
state = IceRestartState::CONNECTED;
@@ -685,9 +684,9 @@
} else {
state = IceRestartState::DISCONNECTED;
}
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IceRestartState",
- static_cast<int>(state),
- static_cast<int>(IceRestartState::MAX_VALUE));
+ metrics_observer_->IncrementEnumCounter(
+ webrtc::kEnumCounterIceRestart, static_cast<int>(state),
+ static_cast<int>(IceRestartState::MAX_VALUE));
}
// Time for a new allocator.
diff --git a/p2p/base/p2ptransportchannel_unittest.cc b/p2p/base/p2ptransportchannel_unittest.cc
index 1c3ae9a..cbf8b25 100644
--- a/p2p/base/p2ptransportchannel_unittest.cc
+++ b/p2p/base/p2ptransportchannel_unittest.cc
@@ -13,6 +13,7 @@
#include <memory>
#include "absl/memory/memory.h"
+#include "api/fakemetricsobserver.h"
#include "p2p/base/fakeportallocator.h"
#include "p2p/base/icetransportinternal.h"
#include "p2p/base/p2ptransportchannel.h"
@@ -36,7 +37,6 @@
#include "rtc_base/ssladapter.h"
#include "rtc_base/thread.h"
#include "rtc_base/virtualsocketserver.h"
-#include "system_wrappers/include/metrics_default.h"
namespace {
@@ -207,10 +207,15 @@
ep1_.allocator_.reset(
CreateBasicPortAllocator(&ep1_.network_manager_, stun_servers,
kTurnUdpIntAddr, rtc::SocketAddress()));
+ ep1_.metrics_observer_ =
+ new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+ ep1_.allocator_->SetMetricsObserver(ep1_.metrics_observer_);
ep2_.allocator_.reset(
CreateBasicPortAllocator(&ep2_.network_manager_, stun_servers,
kTurnUdpIntAddr, rtc::SocketAddress()));
- webrtc::metrics::Reset();
+ ep2_.metrics_observer_ =
+ new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
+ ep2_.allocator_->SetMetricsObserver(ep2_.metrics_observer_);
}
protected:
@@ -277,7 +282,7 @@
Candidates candidates;
};
- struct Endpoint : public sigslot::has_slots<> {
+ struct Endpoint {
Endpoint()
: role_(ICEROLE_UNKNOWN),
tiebreaker_(0),
@@ -308,15 +313,10 @@
allocator_->set_allow_tcp_listen(allow_tcp_listen);
}
- void OnIceRegathering(PortAllocatorSession*, IceRegatheringReason reason) {
- ++ice_regathering_counter_[reason];
- }
-
- int GetIceRegatheringCountForReason(IceRegatheringReason reason) {
- return ice_regathering_counter_[reason];
- }
-
rtc::FakeNetworkManager network_manager_;
+ // |metrics_observer_| should outlive |allocator_| as the former may be
+ // used by the latter.
+ rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer_;
std::unique_ptr<BasicPortAllocator> allocator_;
ChannelData cd1_;
ChannelData cd2_;
@@ -326,7 +326,6 @@
bool save_candidates_;
std::vector<std::unique_ptr<CandidatesData>> saved_candidates_;
bool ready_to_send_ = false;
- std::map<IceRegatheringReason, int> ice_regathering_counter_;
};
ChannelData* GetChannelData(rtc::PacketTransportInternal* transport) {
@@ -354,14 +353,12 @@
ice_ep1_cd1_ch, ice_ep2_cd1_ch));
ep2_.cd1_.ch_.reset(CreateChannel(1, ICE_CANDIDATE_COMPONENT_DEFAULT,
ice_ep2_cd1_ch, ice_ep1_cd1_ch));
+ ep1_.cd1_.ch_->SetMetricsObserver(ep1_.metrics_observer_);
+ ep2_.cd1_.ch_->SetMetricsObserver(ep2_.metrics_observer_);
ep1_.cd1_.ch_->SetIceConfig(ep1_config);
ep2_.cd1_.ch_->SetIceConfig(ep2_config);
ep1_.cd1_.ch_->MaybeStartGathering();
ep2_.cd1_.ch_->MaybeStartGathering();
- ep1_.cd1_.ch_->allocator_session()->SignalIceRegathering.connect(
- &ep1_, &Endpoint::OnIceRegathering);
- ep2_.cd1_.ch_->allocator_session()->SignalIceRegathering.connect(
- &ep2_, &Endpoint::OnIceRegathering);
}
void CreateChannels() {
@@ -444,6 +441,9 @@
BasicPortAllocator* GetAllocator(int endpoint) {
return GetEndpoint(endpoint)->allocator_.get();
}
+ webrtc::FakeMetricsObserver* GetMetricsObserver(int endpoint) {
+ return GetEndpoint(endpoint)->metrics_observer_;
+ }
void AddAddress(int endpoint, const SocketAddress& addr) {
GetEndpoint(endpoint)->network_manager_.AddInterface(addr);
}
@@ -1266,15 +1266,15 @@
ep1_ch1()->SetIceParameters(kIceParams[2]);
ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
ep1_ch1()->MaybeStartGathering();
- EXPECT_EQ(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRestartState",
+ EXPECT_EQ(1, GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRestart,
static_cast<int>(IceRestartState::DISCONNECTED)));
ep2_ch1()->SetIceParameters(kIceParams[3]);
ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
ep2_ch1()->MaybeStartGathering();
- EXPECT_EQ(2, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRestartState",
+ EXPECT_EQ(1, GetMetricsObserver(1)->GetEnumCounter(
+ webrtc::kEnumCounterIceRestart,
static_cast<int>(IceRestartState::DISCONNECTED)));
DestroyChannels();
@@ -1295,15 +1295,15 @@
ep1_ch1()->SetIceParameters(kIceParams[2]);
ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
ep1_ch1()->MaybeStartGathering();
- EXPECT_EQ(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRestartState",
+ EXPECT_EQ(1, GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRestart,
static_cast<int>(IceRestartState::CONNECTED)));
ep2_ch1()->SetIceParameters(kIceParams[3]);
ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
ep2_ch1()->MaybeStartGathering();
- EXPECT_EQ(2, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRestartState",
+ EXPECT_EQ(1, GetMetricsObserver(1)->GetEnumCounter(
+ webrtc::kEnumCounterIceRestart,
static_cast<int>(IceRestartState::CONNECTED)));
DestroyChannels();
@@ -1321,15 +1321,15 @@
ep1_ch1()->SetIceParameters(kIceParams[2]);
ep1_ch1()->SetRemoteIceParameters(kIceParams[3]);
ep1_ch1()->MaybeStartGathering();
- EXPECT_EQ(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRestartState",
+ EXPECT_EQ(1, GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRestart,
static_cast<int>(IceRestartState::CONNECTING)));
ep2_ch1()->SetIceParameters(kIceParams[3]);
ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
ep2_ch1()->MaybeStartGathering();
- EXPECT_EQ(2, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRestartState",
+ EXPECT_EQ(1, GetMetricsObserver(1)->GetEnumCounter(
+ webrtc::kEnumCounterIceRestart,
static_cast<int>(IceRestartState::CONNECTING)));
DestroyChannels();
@@ -1355,10 +1355,12 @@
// Adding address in ep1 will trigger continual gathering.
AddAddress(0, kAlternateAddrs[0]);
- EXPECT_EQ_SIMULATED_WAIT(1,
- GetEndpoint(0)->GetIceRegatheringCountForReason(
- IceRegatheringReason::NETWORK_CHANGE),
- kDefaultTimeout, clock);
+ EXPECT_EQ_SIMULATED_WAIT(
+ 1,
+ GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
+ static_cast<int>(IceRegatheringReason::NETWORK_CHANGE)),
+ kDefaultTimeout, clock);
ep2_ch1()->SetIceParameters(kIceParams[3]);
ep2_ch1()->SetRemoteIceParameters(kIceParams[2]);
@@ -1367,8 +1369,9 @@
AddAddress(1, kAlternateAddrs[1]);
SIMULATED_WAIT(false, kDefaultTimeout, clock);
// ep2 has not enabled continual gathering.
- EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
- IceRegatheringReason::NETWORK_CHANGE));
+ EXPECT_EQ(0, GetMetricsObserver(1)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
+ static_cast<int>(IceRegatheringReason::NETWORK_CHANGE)));
DestroyChannels();
}
@@ -1396,13 +1399,12 @@
// Timeout value such that all connections are deleted.
const int kNetworkFailureTimeout = 35000;
SIMULATED_WAIT(false, kNetworkFailureTimeout, clock);
- EXPECT_LE(1, GetEndpoint(0)->GetIceRegatheringCountForReason(
- IceRegatheringReason::NETWORK_FAILURE));
- EXPECT_LE(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRegatheringReason",
+ EXPECT_LE(1, GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
static_cast<int>(IceRegatheringReason::NETWORK_FAILURE)));
- EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
- IceRegatheringReason::NETWORK_FAILURE));
+ EXPECT_EQ(0, GetMetricsObserver(1)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
+ static_cast<int>(IceRegatheringReason::NETWORK_FAILURE)));
DestroyChannels();
}
@@ -1431,14 +1433,13 @@
const int kNetworkGatherDuration = 11000;
SIMULATED_WAIT(false, kNetworkGatherDuration, clock);
// Expect regathering to happen 5 times in 11s with 2s interval.
- EXPECT_LE(5, GetEndpoint(0)->GetIceRegatheringCountForReason(
- IceRegatheringReason::OCCASIONAL_REFRESH));
- EXPECT_LE(5, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRegatheringReason",
+ EXPECT_LE(5, GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
static_cast<int>(IceRegatheringReason::OCCASIONAL_REFRESH)));
// Expect no regathering if continual gathering not configured.
- EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
- IceRegatheringReason::OCCASIONAL_REFRESH));
+ EXPECT_EQ(0, GetMetricsObserver(1)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
+ static_cast<int>(IceRegatheringReason::OCCASIONAL_REFRESH)));
DestroyChannels();
}
@@ -1483,8 +1484,10 @@
const int kWaitRegather =
kRegatherInterval * kNumRegathers + kRegatherInterval / 2;
SIMULATED_WAIT(false, kWaitRegather, clock);
- EXPECT_EQ(kNumRegathers, GetEndpoint(0)->GetIceRegatheringCountForReason(
- IceRegatheringReason::OCCASIONAL_REFRESH));
+ EXPECT_EQ(kNumRegathers,
+ GetMetricsObserver(0)->GetEnumCounter(
+ webrtc::kEnumCounterIceRegathering,
+ static_cast<int>(IceRegatheringReason::OCCASIONAL_REFRESH)));
const Connection* new_selected = ep1_ch1()->selected_connection();
diff --git a/p2p/base/regatheringcontroller_unittest.cc b/p2p/base/regatheringcontroller_unittest.cc
index 6ae64e8..18b23e0 100644
--- a/p2p/base/regatheringcontroller_unittest.cc
+++ b/p2p/base/regatheringcontroller_unittest.cc
@@ -13,6 +13,7 @@
#include <string>
#include <vector>
+#include "api/fakemetricsobserver.h"
#include "p2p/base/fakeportallocator.h"
#include "p2p/base/mockicetransport.h"
#include "p2p/base/p2pconstants.h"
diff --git a/p2p/client/basicportallocator.cc b/p2p/client/basicportallocator.cc
index eb14a37..4335dda 100644
--- a/p2p/client/basicportallocator.cc
+++ b/p2p/client/basicportallocator.cc
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "api/umametrics.h"
#include "p2p/base/basicpacketsocketfactory.h"
#include "p2p/base/port.h"
#include "p2p/base/relayport.h"
@@ -27,7 +28,6 @@
#include "rtc_base/helpers.h"
#include "rtc_base/ipaddress.h"
#include "rtc_base/logging.h"
-#include "system_wrappers/include/metrics.h"
using rtc::CreateRandomId;
@@ -196,6 +196,9 @@
void BasicPortAllocator::OnIceRegathering(PortAllocatorSession* session,
IceRegatheringReason reason) {
+ if (!metrics_observer()) {
+ return;
+ }
// If the session has not been taken by an active channel, do not report the
// metric.
for (auto& allocator_session : pooled_sessions()) {
@@ -204,9 +207,9 @@
}
}
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IceRegatheringReason",
- static_cast<int>(reason),
- static_cast<int>(IceRegatheringReason::MAX_VALUE));
+ metrics_observer()->IncrementEnumCounter(
+ webrtc::kEnumCounterIceRegathering, static_cast<int>(reason),
+ static_cast<int>(IceRegatheringReason::MAX_VALUE));
}
BasicPortAllocator::~BasicPortAllocator() {
diff --git a/p2p/client/basicportallocator_unittest.cc b/p2p/client/basicportallocator_unittest.cc
index b3e52ff..d995eb1 100644
--- a/p2p/client/basicportallocator_unittest.cc
+++ b/p2p/client/basicportallocator_unittest.cc
@@ -34,7 +34,6 @@
#include "rtc_base/ssladapter.h"
#include "rtc_base/thread.h"
#include "rtc_base/virtualsocketserver.h"
-#include "system_wrappers/include/metrics_default.h"
using rtc::IPAddress;
using rtc::SocketAddress;
@@ -169,7 +168,6 @@
kRelaySslTcpIntAddr));
allocator_->Initialize();
allocator_->set_step_delay(kMinimumStepDelay);
- webrtc::metrics::Reset();
}
void AddInterface(const SocketAddress& addr) {
@@ -2370,21 +2368,4 @@
expected_stun_keepalive_interval);
}
-TEST_F(BasicPortAllocatorTest, IceRegatheringMetricsLoggedWhenNetworkChanges) {
- // Only test local ports to simplify test.
- ResetWithNoServersOrNat();
- AddInterface(kClientAddr, "test_net0");
- ASSERT_TRUE(CreateSession(ICE_CANDIDATE_COMPONENT_RTP));
- session_->StartGettingPorts();
- EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
- kDefaultAllocationTimeout, fake_clock);
- candidate_allocation_done_ = false;
- AddInterface(kClientAddr2, "test_net1");
- EXPECT_TRUE_SIMULATED_WAIT(candidate_allocation_done_,
- kDefaultAllocationTimeout, fake_clock);
- EXPECT_EQ(1, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.IceRegatheringReason",
- static_cast<int>(IceRegatheringReason::NETWORK_CHANGE)));
-}
-
} // namespace cricket
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
index 329d667..195d185 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -206,7 +206,6 @@
"../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",
]
@@ -297,6 +296,7 @@
":rtc_pc",
":rtc_pc_base",
"../api:array_view",
+ "../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../call:rtp_interfaces",
"../logging:rtc_event_log_api",
@@ -334,6 +334,7 @@
]
deps = [
":pc_test_utils",
+ "../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
@@ -493,6 +494,7 @@
":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 c560e05..ebf7cee 100644
--- a/pc/peerconnection.cc
+++ b/pc/peerconnection.cc
@@ -52,7 +52,6 @@
#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;
@@ -384,10 +383,15 @@
return desc1.contents().size() == desc2.contents().size();
}
-void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
- cricket::MediaType media_type) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocol", protocol_type,
- kEnumCounterKeyProtocolMax);
+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);
static const std::map<std::pair<KeyExchangeProtocolType, cricket::MediaType>,
KeyExchangeProtocolMedia>
proto_media_counter_map = {
@@ -406,8 +410,9 @@
auto it = proto_media_counter_map.find({protocol_type, media_type});
if (it != proto_media_counter_map.end()) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.KeyProtocolByMedia",
- it->second, kEnumCounterKeyProtocolMediaTypeMax);
+ uma_observer->IncrementEnumCounter(webrtc::kEnumCounterKeyProtocolMediaType,
+ it->second,
+ kEnumCounterKeyProtocolMediaTypeMax);
}
}
@@ -417,7 +422,9 @@
// 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) {
+RTCError VerifyCrypto(const SessionDescription* desc,
+ bool dtls_enabled,
+ rtc::scoped_refptr<webrtc::UMAObserver> uma_observer) {
const cricket::ContentGroup* bundle =
desc->GetGroupByName(cricket::GROUP_TYPE_BUNDLE);
for (const cricket::ContentInfo& content_info : desc->contents()) {
@@ -427,7 +434,8 @@
// Note what media is used with each crypto protocol, for all sections.
NoteKeyProtocolAndMedia(dtls_enabled ? webrtc::kEnumCounterKeyProtocolDtls
: webrtc::kEnumCounterKeyProtocolSdes,
- content_info.media_description()->type());
+ content_info.media_description()->type(),
+ uma_observer);
const std::string& mid = content_info.name;
if (bundle && bundle->HasContentName(mid) &&
mid != *(bundle->FirstContentName())) {
@@ -931,16 +939,6 @@
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
@@ -3066,6 +3064,18 @@
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) {
@@ -5293,7 +5303,9 @@
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
- ReportTransportStats();
+ if (metrics_observer()) {
+ ReportTransportStats();
+ }
break;
default:
RTC_NOTREACHED();
@@ -5345,9 +5357,11 @@
void PeerConnection::OnTransportControllerDtlsHandshakeError(
rtc::SSLHandshakeError error) {
- RTC_HISTOGRAM_ENUMERATION(
- "WebRTC.PeerConnection.DtlsHandshakeError", static_cast<int>(error),
- static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
+ if (metrics_observer()) {
+ metrics_observer()->IncrementEnumCounter(
+ webrtc::kEnumCounterDtlsHandshakeError, static_cast<int>(error),
+ static_cast<int>(rtc::SSLHandshakeError::MAX_VALUE));
+ }
}
void PeerConnection::EnableSending() {
@@ -5785,7 +5799,8 @@
std::string crypto_error;
if (webrtc_session_desc_factory_->SdesPolicy() == cricket::SEC_REQUIRED ||
dtls_enabled_) {
- RTCError crypto_error = VerifyCrypto(sdesc->description(), dtls_enabled_);
+ RTCError crypto_error =
+ VerifyCrypto(sdesc->description(), dtls_enabled_, uma_observer_);
if (!crypto_error.ok()) {
return crypto_error;
}
@@ -5912,6 +5927,9 @@
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;
@@ -5936,8 +5954,8 @@
} else if (num_audio_tracks > 0 || num_video_tracks > 0) {
format = kSdpFormatReceivedSimple;
}
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SdpFormatReceived", format,
- kSdpFormatReceivedMax);
+ uma_observer_->IncrementEnumCounter(kEnumCounterSdpFormatReceived, format,
+ kSdpFormatReceivedMax);
}
void PeerConnection::NoteUsageEvent(UsageEvent event) {
@@ -5947,33 +5965,42 @@
void PeerConnection::ReportUsagePattern() const {
RTC_DLOG(LS_INFO) << "Usage signature is " << usage_event_accumulator_;
- RTC_HISTOGRAM_ENUMERATION_SPARSE("WebRTC.PeerConnection.UsagePattern",
- usage_event_accumulator_,
- static_cast<int>(UsageEvent::MAX_VALUE));
+ if (uma_observer_) {
+ uma_observer_->IncrementSparseEnumCounter(kEnumCounterUsagePattern,
+ usage_event_accumulator_);
+ }
}
void PeerConnection::ReportNegotiatedSdpSemantics(
const SessionDescriptionInterface& answer) {
- SdpSemanticNegotiated semantics_negotiated;
+ if (!uma_observer_) {
+ return;
+ }
switch (answer.description()->msid_signaling()) {
case 0:
- semantics_negotiated = kSdpSemanticNegotiatedNone;
+ uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
+ kSdpSemanticNegotiatedNone,
+ kSdpSemanticNegotiatedMax);
break;
case cricket::kMsidSignalingMediaSection:
- semantics_negotiated = kSdpSemanticNegotiatedUnifiedPlan;
+ uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
+ kSdpSemanticNegotiatedUnifiedPlan,
+ kSdpSemanticNegotiatedMax);
break;
case cricket::kMsidSignalingSsrcAttribute:
- semantics_negotiated = kSdpSemanticNegotiatedPlanB;
+ uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
+ kSdpSemanticNegotiatedPlanB,
+ kSdpSemanticNegotiatedMax);
break;
case cricket::kMsidSignalingMediaSection |
cricket::kMsidSignalingSsrcAttribute:
- semantics_negotiated = kSdpSemanticNegotiatedMixed;
+ uma_observer_->IncrementEnumCounter(kEnumCounterSdpSemanticNegotiated,
+ kSdpSemanticNegotiatedMixed,
+ kSdpSemanticNegotiatedMax);
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
@@ -6067,6 +6094,7 @@
// 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 :
@@ -6075,6 +6103,7 @@
continue;
}
+ PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
@@ -6082,26 +6111,26 @@
if (local.protocol() == cricket::TCP_PROTOCOL_NAME ||
(local.type() == RELAY_PORT_TYPE &&
local.relay_protocol() == cricket::TCP_PROTOCOL_NAME)) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_TCP",
- GetIceCandidatePairCounter(local, remote),
- kIceCandidatePairMax);
+ type = kEnumCounterIceCandidatePairTypeTcp;
} else if (local.protocol() == cricket::UDP_PROTOCOL_NAME) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.CandidatePairType_UDP",
- GetIceCandidatePairCounter(local, remote),
- kIceCandidatePairMax);
+ type = kEnumCounterIceCandidatePairTypeUdp;
} else {
RTC_CHECK(0);
}
+ metrics_observer()->IncrementEnumCounter(
+ type, GetIceCandidatePairCounter(local, remote),
+ kIceCandidatePairMax);
// Increment the counter for IP type.
if (local.address().family() == AF_INET) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
- kBestConnections_IPv4,
- kPeerConnectionAddressFamilyCounter_Max);
+ metrics_observer()->IncrementEnumCounter(
+ kEnumCounterAddressFamily, kBestConnections_IPv4,
+ kPeerConnectionAddressFamilyCounter_Max);
+
} else if (local.address().family() == AF_INET6) {
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.IPMetrics",
- kBestConnections_IPv6,
- kPeerConnectionAddressFamilyCounter_Max);
+ metrics_observer()->IncrementEnumCounter(
+ kEnumCounterAddressFamily, kBestConnections_IPv6,
+ kPeerConnectionAddressFamilyCounter_Max);
} else {
RTC_CHECK(0);
}
@@ -6114,6 +6143,7 @@
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;
}
@@ -6125,53 +6155,33 @@
return;
}
- 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;
- }
+ 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 (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;
- }
+ 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);
}
}
}
diff --git a/pc/peerconnection.h b/pc/peerconnection.h
index 44b66c3..7d7315d 100644
--- a/pc/peerconnection.h
+++ b/pc/peerconnection.h
@@ -66,8 +66,7 @@
CANDIDATE_COLLECTED = 0x80,
REMOTE_CANDIDATE_ADDED = 0x100,
ICE_STATE_CONNECTED = 0x200,
- CLOSE_CALLED = 0x400,
- MAX_VALUE = 0x800,
+ CLOSE_CALLED = 0x400
};
explicit PeerConnection(PeerConnectionFactory* factory,
diff --git a/pc/peerconnection_histogram_unittest.cc b/pc/peerconnection_histogram_unittest.cc
index 64c4cfe..b998372 100644
--- a/pc/peerconnection_histogram_unittest.cc
+++ b/pc/peerconnection_histogram_unittest.cc
@@ -11,6 +11,7 @@
#include <tuple>
#include "absl/memory/memory.h"
+#include "api/fakemetricsobserver.h"
#include "api/peerconnectionproxy.h"
#include "media/base/fakemediaengine.h"
#include "pc/mediasession.h"
@@ -21,7 +22,6 @@
#include "pc/test/fakesctptransport.h"
#include "rtc_base/gunit.h"
#include "rtc_base/virtualsocketserver.h"
-#include "system_wrappers/include/metrics_default.h"
namespace webrtc {
@@ -127,7 +127,6 @@
PeerConnectionUsageHistogramTest()
: vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
- webrtc::metrics::Reset();
}
WrapperPtr CreatePeerConnection() {
@@ -176,12 +175,14 @@
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(
- 1u == webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"),
- kDefaultTimeout);
- EXPECT_EQ(1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
- expected_fingerprint));
+ ASSERT_TRUE_WAIT(caller_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, expected_fingerprint),
+ kDefaultTimeout);
}
#ifndef WEBRTC_ANDROID
@@ -192,6 +193,9 @@
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");
caller->PrepareToExchangeCandidates(callee.get());
@@ -209,16 +213,19 @@
PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_EQ(2,
- webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
- EXPECT_EQ(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
- expected_fingerprint));
+ EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, expected_fingerprint));
+ EXPECT_TRUE(callee_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, 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");
caller->PrepareToExchangeCandidates(callee.get());
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -233,10 +240,10 @@
PeerConnection::UsageEvent::REMOTE_CANDIDATE_ADDED,
PeerConnection::UsageEvent::ICE_STATE_CONNECTED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_EQ(2,
- webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
- EXPECT_EQ(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
- expected_fingerprint));
+ EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, expected_fingerprint));
+ EXPECT_TRUE(callee_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, expected_fingerprint));
}
#endif // HAVE_SCTP
#endif // WEBRTC_ANDROID
@@ -252,15 +259,14 @@
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_EQ(1,
- webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
- EXPECT_EQ(1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
- expected_fingerprint));
+ EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, expected_fingerprint));
}
TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
@@ -274,6 +280,7 @@
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());
@@ -282,10 +289,8 @@
MakeUsageFingerprint({PeerConnection::UsageEvent::STUN_SERVER_ADDED,
PeerConnection::UsageEvent::TURN_SERVER_ADDED,
PeerConnection::UsageEvent::CLOSE_CALLED});
- EXPECT_EQ(1,
- webrtc::metrics::NumSamples("WebRTC.PeerConnection.UsagePattern"));
- EXPECT_EQ(1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.UsagePattern",
- expected_fingerprint));
+ EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterUsagePattern, expected_fingerprint));
}
} // namespace webrtc
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index 525eba3..ed5db1f 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -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 @@
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
- webrtc::metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
@@ -1514,17 +1513,20 @@
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, 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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 @@
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 |
diff --git a/pc/peerconnection_rtp_unittest.cc b/pc/peerconnection_rtp_unittest.cc
index df1b1ee..50fe0dc 100644
--- a/pc/peerconnection_rtp_unittest.cc
+++ b/pc/peerconnection_rtp_unittest.cc
@@ -17,6 +17,7 @@
#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"
@@ -31,7 +32,6 @@
#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
@@ -77,9 +77,7 @@
CreateBuiltinVideoEncoderFactory(),
CreateBuiltinVideoDecoderFactory(),
nullptr /* audio_mixer */,
- nullptr /* audio_processing */)) {
- webrtc::metrics::Reset();
- }
+ nullptr /* audio_processing */)) {}
std::unique_ptr<PeerConnectionWrapper> CreatePeerConnection() {
return CreatePeerConnection(RTCConfiguration());
@@ -1371,6 +1369,7 @@
caller->AddAudioTrack("caller_audio");
auto callee = CreatePeerConnectionWithUnifiedPlan();
callee->AddAudioTrack("callee_audio");
+ auto caller_observer = caller->RegisterFakeMetricsObserver();
ASSERT_TRUE(caller->ExchangeOfferAnswerWith(callee.get()));
@@ -1385,11 +1384,8 @@
EXPECT_EQ(cricket::kMsidSignalingMediaSection,
answer->description()->msid_signaling());
// Check that this is counted correctly
- EXPECT_EQ(2, webrtc::metrics::NumSamples(
- "WebRTC.PeerConnection.SdpSemanticNegotiated"));
- EXPECT_EQ(2, webrtc::metrics::NumEvents(
- "WebRTC.PeerConnection.SdpSemanticNegotiated",
- kSdpSemanticNegotiatedUnifiedPlan));
+ EXPECT_TRUE(caller_observer->ExpectOnlySingleEnumCount(
+ kEnumCounterSdpSemanticNegotiated, kSdpSemanticNegotiatedUnifiedPlan));
}
TEST_F(PeerConnectionMsidSignalingTest, PlanBOfferToUnifiedPlanAnswer) {
@@ -1474,14 +1470,12 @@
auto caller = CreatePeerConnectionWithUnifiedPlan();
caller->CreateDataChannel("dc");
auto callee = CreatePeerConnectionWithUnifiedPlan();
+ auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
- // 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));
+
+ EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
+ kEnumCounterSdpFormatReceived, kSdpFormatReceivedNoTracks));
}
#endif // HAVE_SCTP
@@ -1490,28 +1484,24 @@
caller->AddAudioTrack("audio");
caller->AddVideoTrack("video");
auto callee = CreatePeerConnectionWithPlanB();
+ auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
- // 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));
+
+ EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
+ kEnumCounterSdpFormatReceived, 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_EQ(1, webrtc::metrics::NumSamples(
- "WebRTC.PeerConnection.SdpFormatReceived"));
- EXPECT_EQ(
- 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SdpFormatReceived",
- kSdpFormatReceivedSimple));
+ EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
+ kEnumCounterSdpFormatReceived, kSdpFormatReceivedSimple));
}
TEST_F(SdpFormatReceivedTest, ComplexUnifiedIsReportedAsComplexUnifiedPlan) {
@@ -1520,14 +1510,12 @@
caller->AddAudioTrack("audio2");
caller->AddVideoTrack("video");
auto callee = CreatePeerConnectionWithPlanB();
+ auto callee_metrics = callee->RegisterFakeMetricsObserver();
ASSERT_TRUE(callee->SetRemoteDescription(caller->CreateOffer()));
- // 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));
+
+ EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
+ kEnumCounterSdpFormatReceived, kSdpFormatReceivedComplexUnifiedPlan));
}
TEST_F(SdpFormatReceivedTest, ComplexPlanBIsReportedAsComplexPlanB) {
@@ -1535,17 +1523,15 @@
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()));
- // 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));
+
+ EXPECT_TRUE(callee_metrics->ExpectOnlySingleEnumCount(
+ kEnumCounterSdpFormatReceived, kSdpFormatReceivedComplexPlanB));
}
// Sender setups in a call.
diff --git a/pc/peerconnectionwrapper.cc b/pc/peerconnectionwrapper.cc
index 5006337..0d19cf3 100644
--- a/pc/peerconnectionwrapper.cc
+++ b/pc/peerconnectionwrapper.cc
@@ -320,4 +320,12 @@
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 436460e..f7de67e 100644
--- a/pc/peerconnectionwrapper.h
+++ b/pc/peerconnectionwrapper.h
@@ -16,6 +16,7 @@
#include <string>
#include <vector>
+#include "api/fakemetricsobserver.h"
#include "api/peerconnectioninterface.h"
#include "pc/test/mockpeerconnectionobservers.h"
#include "rtc_base/function_view.h"
@@ -170,6 +171,10 @@
// 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,
@@ -180,6 +185,7 @@
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 28349ad..3b33f85 100644
--- a/pc/srtpsession.cc
+++ b/pc/srtpsession.cc
@@ -139,7 +139,11 @@
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;
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtpUnprotectError",
+ if (metrics_observer_) {
+ metrics_observer_->IncrementSparseEnumCounter(
+ webrtc::kEnumCounterSrtpUnprotectError, err);
+ }
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.UnprotectSrtpError",
static_cast<int>(err), kSrtpErrorCodeBoundary);
return false;
}
@@ -157,7 +161,11 @@
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;
- RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.SrtcpUnprotectError",
+ if (metrics_observer_) {
+ metrics_observer_->IncrementSparseEnumCounter(
+ webrtc::kEnumCounterSrtcpUnprotectError, err);
+ }
+ RTC_HISTOGRAM_ENUMERATION("WebRTC.PeerConnection.UnprotectSrtcpError",
static_cast<int>(err), kSrtpErrorCodeBoundary);
return false;
}
diff --git a/pc/srtpsession_unittest.cc b/pc/srtpsession_unittest.cc
index b1bc9f0..66e1cea 100644
--- a/pc/srtpsession_unittest.cc
+++ b/pc/srtpsession_unittest.cc
@@ -13,21 +13,20 @@
#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);
@@ -137,6 +136,9 @@
// 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));
@@ -147,38 +149,29 @@
rtp_packet_[0] = 0x12;
rtcp_packet_[1] = 0x34;
EXPECT_FALSE(s2_.UnprotectRtp(rtp_packet_, rtp_len_, &out_len));
- EXPECT_EQ(1, webrtc::metrics::NumSamples(
- "WebRTC.PeerConnection.SrtpUnprotectError"));
- EXPECT_EQ(
- 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
- srtp_err_status_bad_param));
+ EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterSrtpUnprotectError, srtp_err_status_bad_param));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
- EXPECT_EQ(1, webrtc::metrics::NumSamples(
- "WebRTC.PeerConnection.SrtcpUnprotectError"));
- EXPECT_EQ(
- 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
- srtp_err_status_auth_fail));
+ EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterSrtcpUnprotectError, 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_EQ(1, webrtc::metrics::NumSamples(
- "WebRTC.PeerConnection.SrtpUnprotectError"));
- EXPECT_EQ(
- 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtpUnprotectError",
- srtp_err_status_auth_fail));
+ EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterSrtpUnprotectError, srtp_err_status_auth_fail));
EXPECT_FALSE(s2_.UnprotectRtcp(rtcp_packet_, rtcp_len_, &out_len));
- EXPECT_EQ(1, webrtc::metrics::NumSamples(
- "WebRTC.PeerConnection.SrtcpUnprotectError"));
- EXPECT_EQ(
- 1, webrtc::metrics::NumEvents("WebRTC.PeerConnection.SrtcpUnprotectError",
- srtp_err_status_cant_check));
+ EXPECT_TRUE(metrics_observer->ExpectOnlySingleEnumCount(
+ webrtc::kEnumCounterSrtcpUnprotectError, srtp_err_status_cant_check));
}
// Test that we fail when using buffers that are too small.
diff --git a/rtc_base/BUILD.gn b/rtc_base/BUILD.gn
index 18d6897..4f4cdf1 100644
--- a/rtc_base/BUILD.gn
+++ b/rtc_base/BUILD.gn
@@ -1066,7 +1066,6 @@
":rtc_base_approved",
":rtc_base_tests_utils",
"../system_wrappers:field_trial_default",
- "../system_wrappers:metrics_default",
"../test:field_trial",
"../test:fileutils",
"../test:test_support",
diff --git a/rtc_base/sslstreamadapter.h b/rtc_base/sslstreamadapter.h
index 2d4e19f..827dc45 100644
--- a/rtc_base/sslstreamadapter.h
+++ b/rtc_base/sslstreamadapter.h
@@ -22,7 +22,6 @@
// Constants for SSL profile.
const int TLS_NULL_WITH_NULL_NULL = 0;
-const int SSL_CIPHER_SUITE_MAX_VALUE = 0xFFFF;
// Constants for SRTP profiles.
const int SRTP_INVALID_CRYPTO_SUITE = 0;
@@ -38,7 +37,6 @@
#ifndef SRTP_AEAD_AES_256_GCM
const int SRTP_AEAD_AES_256_GCM = 0x0008;
#endif
-const int SRTP_CRYPTO_SUITE_MAX_VALUE = 0xFFFF;
// Names of SRTP profiles listed above.
// 128-bit AES with 80-bit SHA-1 HMAC.
diff --git a/rtc_base/unittest_main.cc b/rtc_base/unittest_main.cc
index 0b5a39d..df9622b 100644
--- a/rtc_base/unittest_main.cc
+++ b/rtc_base/unittest_main.cc
@@ -20,7 +20,6 @@
#include "rtc_base/ssladapter.h"
#include "rtc_base/sslstreamadapter.h"
#include "system_wrappers/include/field_trial_default.h"
-#include "system_wrappers/include/metrics_default.h"
#include "test/field_trial.h"
#include "test/testsupport/fileutils.h"
@@ -82,7 +81,6 @@
// InitFieldTrialsFromString stores the char*, so the char array must outlive
// the application.
webrtc::field_trial::InitFieldTrialsFromString(FLAG_force_fieldtrials);
- webrtc::metrics::Enable();
#if defined(WEBRTC_WIN)
if (!FLAG_default_error_handlers) {
diff --git a/system_wrappers/include/metrics.h b/system_wrappers/include/metrics.h
index 99b8194..13ed2c9 100644
--- a/system_wrappers/include/metrics.h
+++ b/system_wrappers/include/metrics.h
@@ -127,9 +127,6 @@
// Histogram for enumerators (evenly spaced buckets).
// |boundary| should be above the max enumerator sample.
-//
-// TODO(qingsi): Refactor the default implementation given by RtcHistogram,
-// which is already sparse, and remove the boundary argument from the macro.
#define RTC_HISTOGRAM_ENUMERATION_SPARSE(name, sample, boundary) \
RTC_HISTOGRAM_COMMON_BLOCK( \
name, sample, \