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

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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}
This commit is contained in:
Qingsi Wang 2018-07-11 18:33:52 +00:00 committed by Commit Bot
parent 8da935e2c9
commit 78fef76e6a
25 changed files with 510 additions and 308 deletions
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21
api/BUILD.gn
View File

@ -86,6 +86,7 @@ rtc_static_library("libjingle_peerconnection_api") {
"statstypes.cc",
"statstypes.h",
"turncustomizer.h",
"umametrics.cc",
"umametrics.h",
"videosourceproxy.h",
]
@ -451,6 +452,26 @@ if (rtc_include_tests) {
]
}
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

87
api/fakemetricsobserver.cc Normal file
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@ -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

56
api/fakemetricsobserver.h Normal file
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@ -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_

21
api/umametrics.cc Normal file
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@ -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

4
api/umametrics.h
View File

@ -176,13 +176,13 @@ class MetricsObserverInterface : public rtc::RefCountInterface {
// 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;

3
p2p/BUILD.gn
View File

@ -91,7 +91,6 @@ rtc_static_library("rtc_p2p") {
"../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 @@ if (rtc_include_tests) {
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",

9
p2p/base/p2ptransportchannel.cc
View File

@ -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 @@ void P2PTransportChannel::MaybeStartGathering() {
SignalGatheringState(this);
}
if (!allocator_sessions_.empty()) {
if (metrics_observer_ && !allocator_sessions_.empty()) {
IceRestartState state;
if (writable()) {
state = IceRestartState::CONNECTED;
@ -685,9 +684,9 @@ void P2PTransportChannel::MaybeStartGathering() {
} 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.

99
p2p/base/p2ptransportchannel_unittest.cc
View File

@ -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 @@ class P2PTransportChannelTestBase : public testing::Test,
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 @@ class P2PTransportChannelTestBase : public testing::Test,
Candidates candidates;
};
struct Endpoint : public sigslot::has_slots<> {
struct Endpoint {
Endpoint()
: role_(ICEROLE_UNKNOWN),
tiebreaker_(0),
@ -308,15 +313,10 @@ class P2PTransportChannelTestBase : public testing::Test,
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 @@ class P2PTransportChannelTestBase : public testing::Test,
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 @@ class P2PTransportChannelTestBase : public testing::Test,
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 @@ class P2PTransportChannelTestBase : public testing::Test,
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 @@ TEST_F(P2PTransportChannelTest, TestUMAIceRestartWhileDisconnected) {
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 @@ TEST_F(P2PTransportChannelTest, TestUMAIceRestartWhileConnected) {
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 @@ TEST_F(P2PTransportChannelTest, TestUMAIceRestartWhileConnecting) {
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 @@ TEST_F(P2PTransportChannelTest,
// 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 @@ TEST_F(P2PTransportChannelTest,
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 @@ TEST_F(P2PTransportChannelTest,
// 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, GetMetricsObserver(1)->GetEnumCounter(
webrtc::kEnumCounterIceRegathering,
static_cast<int>(IceRegatheringReason::NETWORK_FAILURE)));
EXPECT_EQ(0, GetEndpoint(1)->GetIceRegatheringCountForReason(
IceRegatheringReason::NETWORK_FAILURE));
DestroyChannels();
}
@ -1431,14 +1433,13 @@ TEST_F(P2PTransportChannelTest, TestIceRegatherOnAllNetworksContinual) {
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 @@ class P2PTransportRegatherAllNetworksTest : public P2PTransportChannelTest {
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();

1
p2p/base/regatheringcontroller_unittest.cc
View File

@ -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"

11
p2p/client/basicportallocator.cc
View File

@ -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::Construct() {
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 @@ void BasicPortAllocator::OnIceRegathering(PortAllocatorSession* session,
}
}
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() {

19
p2p/client/basicportallocator_unittest.cc
View File

@ -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 @@ class BasicPortAllocatorTestBase : public testing::Test,
kRelaySslTcpIntAddr));
allocator_->Initialize();
allocator_->set_step_delay(kMinimumStepDelay);
webrtc::metrics::Reset();
}
void AddInterface(const SocketAddress& addr) {
@ -2370,21 +2368,4 @@ TEST_F(BasicPortAllocatorTest,
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

4
pc/BUILD.gn
View File

@ -206,7 +206,6 @@ rtc_static_library("peerconnection") {
"../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 @@ if (rtc_include_tests) {
":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 @@ if (rtc_include_tests) {
]
deps = [
":pc_test_utils",
"../api:fakemetricsobserver",
"../api:libjingle_peerconnection_api",
"../api:libjingle_peerconnection_test_api",
"../api:rtc_stats_api",
@ -493,6 +494,7 @@ if (rtc_include_tests) {
":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",

198
pc/peerconnection.cc
View File

@ -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 @@ bool MediaSectionsHaveSameCount(const SessionDescription& desc1,
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 @@ void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
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 @@ void NoteKeyProtocolAndMedia(KeyExchangeProtocolType protocol_type,
// 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 @@ RTCError VerifyCrypto(const SessionDescription* desc, bool dtls_enabled) {
// 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 @@ bool PeerConnection::Initialize(
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 @@ void PeerConnection::RegisterUMAObserver(UMAObserver* observer) {
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 @@ void PeerConnection::OnTransportControllerConnectionState(
}
SetIceConnectionState(PeerConnectionInterface::kIceConnectionCompleted);
NoteUsageEvent(UsageEvent::ICE_STATE_CONNECTED);
ReportTransportStats();
if (metrics_observer()) {
ReportTransportStats();
}
break;
default:
RTC_NOTREACHED();
@ -5345,9 +5357,11 @@ void PeerConnection::OnTransportControllerCandidatesRemoved(
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 @@ RTCError PeerConnection::ValidateSessionDescription(
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 @@ std::string PeerConnection::GetSessionErrorMsg() {
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 @@ void PeerConnection::ReportSdpFormatReceived(
} 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::NoteUsageEvent(UsageEvent event) {
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 @@ void PeerConnection::ReportTransportStats() {
// 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 @@ void PeerConnection::ReportBestConnectionState(
continue;
}
PeerConnectionEnumCounterType type = kPeerConnectionEnumCounterMax;
const cricket::Candidate& local = connection_info.local_candidate;
const cricket::Candidate& remote = connection_info.remote_candidate;
@ -6082,26 +6111,26 @@ void PeerConnection::ReportBestConnectionState(
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::ReportBestConnectionState(
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 @@ void PeerConnection::ReportNegotiatedCiphers(
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);
}
}
}

3
pc/peerconnection.h
View File

@ -66,8 +66,7 @@ class PeerConnection : public PeerConnectionInternal,
CANDIDATE_COLLECTED = 0x80,
REMOTE_CANDIDATE_ADDED = 0x100,
ICE_STATE_CONNECTED = 0x200,
CLOSE_CALLED = 0x400,
MAX_VALUE = 0x800,
CLOSE_CALLED = 0x400
};
explicit PeerConnection(PeerConnectionFactory* factory,

51
pc/peerconnection_histogram_unittest.cc
View File

@ -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 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
PeerConnectionUsageHistogramTest()
: vss_(new rtc::VirtualSocketServer()), main_(vss_.get()) {
webrtc::metrics::Reset();
}
WrapperPtr CreatePeerConnection() {
@ -176,12 +175,14 @@ class PeerConnectionUsageHistogramTest : public ::testing::Test {
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, UsageFingerprintHistogramFromTimeout) {
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 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintAudioVideo) {
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 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintDataOnly) {
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 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurn) {
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 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
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 @@ TEST_F(PeerConnectionUsageHistogramTest, FingerprintStunTurnInReconfiguration) {
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

94
pc/peerconnection_integrationtest.cc
View File

@ -24,6 +24,7 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/fakemetricsobserver.h"
#include "api/mediastreaminterface.h"
#include "api/peerconnectioninterface.h"
#include "api/peerconnectionproxy.h"
@ -62,7 +63,6 @@
#include "rtc_base/numerics/safe_conversions.h"
#include "rtc_base/testcertificateverifier.h"
#include "rtc_base/virtualsocketserver.h"
#include "system_wrappers/include/metrics_default.h"
#include "test/gmock.h"
using cricket::ContentInfo;
@ -1106,7 +1106,6 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
worker_thread_->SetName("PCWorkerThread", this);
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
webrtc::metrics::Reset();
}
~PeerConnectionIntegrationBaseTest() {
@ -1514,17 +1513,20 @@ class PeerConnectionIntegrationBaseTest : public testing::Test {
int expected_cipher_suite) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
callee_options));
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
ConnectFakeSignaling();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(expected_cipher_suite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
expected_cipher_suite));
EXPECT_EQ(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
expected_cipher_suite));
caller()->pc()->RegisterUMAObserver(nullptr);
}
void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
@ -1694,6 +1696,9 @@ TEST_P(PeerConnectionIntegrationTest, DtmfSenderObserver) {
TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
ASSERT_TRUE(CreatePeerConnectionWrappers());
ConnectFakeSignaling();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
// Do normal offer/answer and wait for some frames to be received in each
// direction.
@ -1704,10 +1709,12 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithDtls) {
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_LE(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_EQ(
0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolSdes));
}
// Uses SDES instead of DTLS for key agreement.
@ -1716,6 +1723,9 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
sdes_config.enable_dtls_srtp.emplace(false);
ASSERT_TRUE(CreatePeerConnectionWrappersWithConfig(sdes_config, sdes_config));
ConnectFakeSignaling();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
// Do normal offer/answer and wait for some frames to be received in each
// direction.
@ -1726,10 +1736,12 @@ TEST_P(PeerConnectionIntegrationTest, EndToEndCallWithSdes) {
MediaExpectations media_expectations;
media_expectations.ExpectBidirectionalAudioAndVideo();
ASSERT_TRUE(ExpectNewFrames(media_expectations));
EXPECT_LE(2, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_EQ(0, webrtc::metrics::NumEvents("WebRTC.PeerConnection.KeyProtocol",
webrtc::kEnumCounterKeyProtocolDtls));
EXPECT_LE(
1, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolSdes));
EXPECT_EQ(
0, caller_observer->GetEnumCounter(webrtc::kEnumCounterKeyProtocol,
webrtc::kEnumCounterKeyProtocolDtls));
}
// Tests that the GetRemoteAudioSSLCertificate method returns the remote DTLS
@ -2731,19 +2743,22 @@ TEST_P(PeerConnectionIntegrationTest, Dtls10CipherStatsAndUmaMetrics) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_10_options,
dtls_10_options));
ConnectFakeSignaling();
// Register UMA observer before signaling begins.
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
EXPECT_EQ(1,
caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test getting cipher stats and UMA metrics when DTLS 1.2 is negotiated.
@ -2753,19 +2768,22 @@ TEST_P(PeerConnectionIntegrationTest, Dtls12CipherStatsAndUmaMetrics) {
ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(dtls_12_options,
dtls_12_options));
ConnectFakeSignaling();
// Register UMA observer before signaling begins.
rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>();
caller()->pc()->RegisterUMAObserver(caller_observer);
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(DtlsConnected(), kDefaultTimeout);
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
EXPECT_TRUE_WAIT(rtc::SSLStreamAdapter::IsAcceptableCipher(
caller()->OldGetStats()->DtlsCipher(), rtc::KT_DEFAULT),
kDefaultTimeout);
EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
caller()->OldGetStats()->SrtpCipher(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.SrtpCryptoSuite.Audio",
kDefaultSrtpCryptoSuite));
EXPECT_EQ(1,
caller_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 can be used if the caller supports DTLS 1.2 and the
@ -3484,15 +3502,19 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
SetUpNetworkInterfaces();
caller()->AddAudioVideoTracks();
callee()->AddAudioVideoTracks();
rtc::scoped_refptr<webrtc::FakeMetricsObserver> metrics_observer(
new rtc::RefCountedObject<webrtc::FakeMetricsObserver>());
caller()->pc()->RegisterUMAObserver(metrics_observer.get());
caller()->CreateAndSetAndSignalOffer();
ASSERT_TRUE_WAIT(SignalingStateStable(), kDefaultTimeout);
// TODO(bugs.webrtc.org/9456): Fix it.
const int num_best_ipv4 = webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv4);
const int num_best_ipv6 = webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.IPMetrics", webrtc::kBestConnections_IPv6);
const int num_best_ipv4 = metrics_observer->GetEnumCounter(
webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv4);
const int num_best_ipv6 = metrics_observer->GetEnumCounter(
webrtc::kEnumCounterAddressFamily, webrtc::kBestConnections_IPv6);
if (TestIPv6()) {
// When IPv6 is enabled, we should prefer an IPv6 connection over an IPv4
// connection.
@ -3503,12 +3525,12 @@ TEST_P(PeerConnectionIntegrationIceStatesTest, VerifyBestConnection) {
EXPECT_EQ(0, num_best_ipv6);
}
EXPECT_EQ(0, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostHost));
EXPECT_EQ(1, webrtc::metrics::NumEvents(
"WebRTC.PeerConnection.CandidatePairType_UDP",
webrtc::kIceCandidatePairHostPublicHostPublic));
EXPECT_EQ(0, metrics_observer->GetEnumCounter(
webrtc::kEnumCounterIceCandidatePairTypeUdp,
webrtc::kIceCandidatePairHostHost));
EXPECT_EQ(1, metrics_observer->GetEnumCounter(
webrtc::kEnumCounterIceCandidatePairTypeUdp,
webrtc::kIceCandidatePairHostPublicHostPublic));
}
constexpr uint32_t kFlagsIPv4NoStun = cricket::PORTALLOCATOR_DISABLE_TCP |

62
pc/peerconnection_rtp_unittest.cc
View File

@ -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 @@ class PeerConnectionRtpBaseTest : public testing::Test {
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 @@ TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanTalkingToOurself) {
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 @@ TEST_F(PeerConnectionMsidSignalingTest, UnifiedPlanTalkingToOurself) {
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 @@ TEST_F(SdpFormatReceivedTest, DataChannelOnlyIsReportedAsNoTracks) {
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 @@ TEST_F(SdpFormatReceivedTest, SimpleUnifiedPlanIsReportedAsSimple) {
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 @@ TEST_F(SdpFormatReceivedTest, ComplexUnifiedIsReportedAsComplexUnifiedPlan) {
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 @@ TEST_F(SdpFormatReceivedTest, ComplexPlanBIsReportedAsComplexPlanB) {
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.

8
pc/peerconnectionwrapper.cc
View File

@ -320,4 +320,12 @@ PeerConnectionWrapper::GetStats() {
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

6
pc/peerconnectionwrapper.h
View File

@ -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 @@ class PeerConnectionWrapper {
// 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 @@ class PeerConnectionWrapper {
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

12
pc/srtpsession.cc
View File

@ -139,7 +139,11 @@ bool SrtpSession::UnprotectRtp(void* p, int in_len, int* out_len) {
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 @@ bool SrtpSession::UnprotectRtcp(void* p, int in_len, int* out_len) {
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;
}

41
pc/srtpsession_unittest.cc
View File

@ -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_F(SrtpSessionTest, TestGetSendStreamPacketIndex) {
// 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 @@ TEST_F(SrtpSessionTest, TestTamperReject) {
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.

1
rtc_base/BUILD.gn
View File

@ -1066,7 +1066,6 @@ if (rtc_include_tests) {
":rtc_base_approved",
":rtc_base_tests_utils",
"../system_wrappers:field_trial_default",
"../system_wrappers:metrics_default",
"../test:field_trial",
"../test:fileutils",
"../test:test_support",

2
rtc_base/sslstreamadapter.h
View File

@ -22,7 +22,6 @@ namespace rtc {
// 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 @@ const int SRTP_AEAD_AES_128_GCM = 0x0007;
#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.

2
rtc_base/unittest_main.cc
View File

@ -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 @@ int main(int argc, char* argv[]) {
// 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) {

3
system_wrappers/include/metrics.h
View File

@ -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, \