commit e36f39e8bae3b1e11422f293e46bd0fc71b76510
parent 763e5000b7bf81f08d72cb29c93582e47c655858
Author: Dan Baker <dbaker@mozilla.com>
Date: Mon, 1 Dec 2025 16:58:17 -0700
Bug 2000941 - Vendor libwebrtc from 706520bd0c
Upstream commit: https://webrtc.googlesource.com/src/+/706520bd0c997a512e7c0e1cef613fe1ef646115
Use injected clock in pc/test
This means setting up the environment in the test classes and
propogating modified versions to the peer connection with the field
trials required.
Bug: webrtc:42223992
Change-Id: I91355023e5b64ab8e045b2e583e7fb6b539475e9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/408101
Commit-Queue: Evan Shrubsole <eshr@webrtc.org>
Commit-Queue: Tomas Gunnarsson <tommi@webrtc.org>
Reviewed-by: Tomas Gunnarsson <tommi@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#45572}
Diffstat:
13 files changed, 91 insertions(+), 57 deletions(-)
diff --git a/third_party/libwebrtc/README.mozilla.last-vendor b/third_party/libwebrtc/README.mozilla.last-vendor
@@ -1,4 +1,4 @@
# ./mach python dom/media/webrtc/third_party_build/vendor-libwebrtc.py --from-local /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc --commit mozpatches libwebrtc
-libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-01T23:55:14.213672+00:00.
+libwebrtc updated from /Users/danielbaker/elm/.moz-fast-forward/moz-libwebrtc commit mozpatches on 2025-12-01T23:58:02.723130+00:00.
# base of lastest vendoring
-176f169606
+706520bd0c
diff --git a/third_party/libwebrtc/moz-patch-stack/s0102.patch b/third_party/libwebrtc/moz-patch-stack/s0102.patch
@@ -601,7 +601,7 @@ index b7561e53b6..fe7eb57423 100644
import("../../webrtc.gni")
diff --git a/pc/BUILD.gn b/pc/BUILD.gn
-index 696829d6ca..81bd2a6215 100644
+index f43d1b9ba1..2efd33c566 100644
--- a/pc/BUILD.gn
+++ b/pc/BUILD.gn
@@ -30,8 +30,8 @@
diff --git a/third_party/libwebrtc/pc/BUILD.gn b/third_party/libwebrtc/pc/BUILD.gn
@@ -2272,6 +2272,8 @@ if (rtc_include_tests && !build_with_chromium) {
"../api:scoped_refptr",
"../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
+ "../api/environment",
+ "../api/environment:environment_factory",
"../api/test/metrics:global_metrics_logger_and_exporter",
"../api/test/metrics:metric",
"../api/units:time_delta",
@@ -2300,6 +2302,7 @@ if (rtc_include_tests && !build_with_chromium) {
"../rtc_base:threading",
"../rtc_base:timeutils",
"../system_wrappers",
+ "../test:create_test_environment",
"../test:create_test_field_trials",
"../test:test_support",
"../test:wait_until",
@@ -2737,6 +2740,7 @@ if (rtc_include_tests && !build_with_chromium) {
"../api:sequence_checker",
"../api/audio:builtin_audio_processing_builder",
"../api/crypto:options",
+ "../api/environment",
"../api/environment:environment_factory",
"../api/metronome",
"../api/rtc_event_log:rtc_event_log_factory",
@@ -2763,7 +2767,9 @@ if (rtc_include_tests && !build_with_chromium) {
"../rtc_base:task_queue_for_test",
"../rtc_base:threading",
"../rtc_base:timeutils",
+ "../system_wrappers",
"../system_wrappers:metrics",
+ "../test:create_test_environment",
"../test:create_test_field_trials",
"../test:test_support",
"../test:wait_until",
@@ -2858,7 +2864,6 @@ if (rtc_include_tests && !build_with_chromium) {
"../api:dtls_transport_interface",
"../api:field_trials",
"../api:field_trials_view",
- "../api:field_trials_view",
"../api:frame_transformer_interface",
"../api:libjingle_logging_api",
"../api:libjingle_peerconnection_api",
diff --git a/third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_adaptation_integrationtest.cc
@@ -14,6 +14,7 @@
#include "api/adaptation/resource.h"
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
+#include "api/environment/environment.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/peer_connection_interface.h"
@@ -29,8 +30,9 @@
#include "pc/test/peer_connection_test_wrapper.h"
#include "rtc_base/checks.h"
#include "rtc_base/thread.h"
-#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
+#include "system_wrappers/include/clock.h"
+#include "test/create_test_environment.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"
@@ -53,10 +55,11 @@ void Negotiate(scoped_refptr<PeerConnectionTestWrapper> caller,
}
TrackWithPeriodicSource CreateTrackWithPeriodicSource(
+ Clock& clock,
scoped_refptr<PeerConnectionFactoryInterface> factory) {
FakePeriodicVideoSource::Config periodic_track_source_config;
periodic_track_source_config.frame_interval_ms = 100;
- periodic_track_source_config.timestamp_offset_ms = TimeMillis();
+ periodic_track_source_config.timestamp_offset_ms = clock.TimeInMilliseconds();
scoped_refptr<FakePeriodicVideoTrackSource> periodic_track_source =
make_ref_counted<FakePeriodicVideoTrackSource>(
periodic_track_source_config, /* remote */ false);
@@ -68,9 +71,9 @@ TrackWithPeriodicSource CreateTrackWithPeriodicSource(
}
// Triggers overuse and obtains VideoSinkWants. Adaptation processing happens in
-// parallel and this function makes no guarantee that the returnd VideoSinkWants
-// have yet to reflect the overuse signal. Used together with EXPECT_TRUE_WAIT
-// to "spam overuse until a change is observed".
+// parallel and this function makes no guarantee that the returned
+// VideoSinkWants have yet to reflect the overuse signal. Used together with
+// EXPECT_TRUE_WAIT to "spam overuse until a change is observed".
VideoSinkWants TriggerOveruseAndGetSinkWants(
scoped_refptr<FakeResource> fake_resource,
const FakePeriodicVideoSource& source) {
@@ -81,7 +84,8 @@ VideoSinkWants TriggerOveruseAndGetSinkWants(
class PeerConnectionAdaptationIntegrationTest : public ::testing::Test {
public:
PeerConnectionAdaptationIntegrationTest()
- : virtual_socket_server_(),
+ : env_(CreateTestEnvironment()),
+ virtual_socket_server_(),
network_thread_(new Thread(&virtual_socket_server_)),
worker_thread_(Thread::Create()) {
RTC_CHECK(network_thread_->Start());
@@ -91,7 +95,7 @@ class PeerConnectionAdaptationIntegrationTest : public ::testing::Test {
scoped_refptr<PeerConnectionTestWrapper> CreatePcWrapper(const char* name) {
scoped_refptr<PeerConnectionTestWrapper> pc_wrapper =
make_ref_counted<PeerConnectionTestWrapper>(
- name, &virtual_socket_server_, network_thread_.get(),
+ name, env_, &virtual_socket_server_, network_thread_.get(),
worker_thread_.get());
PeerConnectionInterface::RTCConfiguration config;
config.sdp_semantics = SdpSemantics::kUnifiedPlan;
@@ -101,6 +105,7 @@ class PeerConnectionAdaptationIntegrationTest : public ::testing::Test {
}
protected:
+ const Environment env_;
VirtualSocketServer virtual_socket_server_;
std::unique_ptr<Thread> network_thread_;
std::unique_ptr<Thread> worker_thread_;
@@ -114,7 +119,7 @@ TEST_F(PeerConnectionAdaptationIntegrationTest,
// Adding a track and negotiating ensures that a VideoSendStream exists.
TrackWithPeriodicSource track_with_source =
- CreateTrackWithPeriodicSource(caller_wrapper->pc_factory());
+ CreateTrackWithPeriodicSource(env_.clock(), caller_wrapper->pc_factory());
auto sender = caller->AddTrack(track_with_source.track, {}).value();
Negotiate(caller_wrapper, callee_wrapper);
// Prefer degrading resolution.
@@ -150,7 +155,7 @@ TEST_F(PeerConnectionAdaptationIntegrationTest,
// Adding a track and negotiating ensures that a VideoSendStream exists.
TrackWithPeriodicSource track_with_source =
- CreateTrackWithPeriodicSource(caller_wrapper->pc_factory());
+ CreateTrackWithPeriodicSource(env_.clock(), caller_wrapper->pc_factory());
auto sender = caller->AddTrack(track_with_source.track, {}).value();
Negotiate(caller_wrapper, callee_wrapper);
// Prefer degrading resolution.
diff --git a/third_party/libwebrtc/pc/peer_connection_callsetup_perf_tests.cc b/third_party/libwebrtc/pc/peer_connection_callsetup_perf_tests.cc
@@ -21,6 +21,7 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/data_channel_interface.h"
+#include "api/environment/environment.h"
#include "api/jsep.h"
#include "api/make_ref_counted.h"
#include "api/rtc_error.h"
@@ -37,6 +38,7 @@
#include "rtc_base/thread.h"
#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
+#include "test/create_test_environment.h"
#include "test/create_test_field_trials.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -60,7 +62,8 @@ class PeerConnectionDataChannelOpenTest
/*dtls_role=*/ConnectionRole>> {
public:
PeerConnectionDataChannelOpenTest()
- : background_thread_(std::make_unique<Thread>(&vss_)) {
+ : env_(CreateTestEnvironment()),
+ background_thread_(std::make_unique<Thread>(&vss_)) {
RTC_CHECK(background_thread_->Start());
// Delay is set to 50ms so we get a 100ms RTT.
vss_.set_delay_mean(/*delay_mean=*/50);
@@ -70,7 +73,7 @@ class PeerConnectionDataChannelOpenTest
scoped_refptr<PeerConnectionTestWrapper> CreatePc(
absl::string_view field_trials = "") {
auto pc_wrapper = make_ref_counted<PeerConnectionTestWrapper>(
- "pc", &vss_, background_thread_.get(), background_thread_.get());
+ "pc", env_, &vss_, background_thread_.get(), background_thread_.get());
pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(),
CreateTestFieldTrialsPtr(field_trials));
@@ -171,6 +174,7 @@ class PeerConnectionDataChannelOpenTest
return true;
}
+ const Environment env_;
VirtualSocketServer vss_;
std::unique_ptr<Thread> background_thread_;
};
diff --git a/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_encodings_integrationtest.cc
@@ -26,6 +26,7 @@
#include "api/audio_codecs/builtin_audio_decoder_factory.h"
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
+#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/jsep.h"
#include "api/make_ref_counted.h"
@@ -57,6 +58,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/physical_socket_server.h"
#include "rtc_base/thread.h"
+#include "test/create_test_environment.h"
#include "test/create_test_field_trials.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -226,14 +228,15 @@ const RTCOutboundRtpStreamStats* FindOutboundRtpByRid(
class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
public:
PeerConnectionEncodingsIntegrationTest()
- : background_thread_(std::make_unique<Thread>(&pss_)) {
+ : env_(CreateTestEnvironment()),
+ background_thread_(std::make_unique<Thread>(&pss_)) {
RTC_CHECK(background_thread_->Start());
}
scoped_refptr<PeerConnectionTestWrapper> CreatePc(
absl::string_view field_trials = "") {
auto pc_wrapper = make_ref_counted<PeerConnectionTestWrapper>(
- "pc", &pss_, background_thread_.get(), background_thread_.get());
+ "pc", env_, &pss_, background_thread_.get(), background_thread_.get());
pc_wrapper->CreatePc({}, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(),
CreateTestFieldTrialsPtr(field_trials));
@@ -419,6 +422,7 @@ class PeerConnectionEncodingsIntegrationTest : public ::testing::Test {
return true;
}
+ const Environment env_;
PhysicalSocketServer pss_;
std::unique_ptr<Thread> background_thread_;
};
@@ -3082,7 +3086,7 @@ class PeerConnectionEncodingsFakeCodecsIntegrationTest
std::make_unique<FakeWebRtcVideoDecoderFactory>();
video_decoder_factory->AddSupportedVideoCodecType("H265");
auto pc_wrapper = make_ref_counted<PeerConnectionTestWrapper>(
- "pc", &pss_, background_thread_.get(), background_thread_.get());
+ "pc", env_, &pss_, background_thread_.get(), background_thread_.get());
pc_wrapper->CreatePc(
{}, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(), std::move(video_encoder_factory),
@@ -3120,7 +3124,7 @@ class PeerConnectionEncodingsFakeCodecsIntegrationTest
{"profile-level-id", "f4001f"}}, // recvonly
{ScalabilityMode::kL1T1}));
auto pc_wrapper = make_ref_counted<PeerConnectionTestWrapper>(
- "pc", &pss_, background_thread_.get(), background_thread_.get());
+ "pc", env_, &pss_, background_thread_.get(), background_thread_.get());
pc_wrapper->CreatePc(
{}, CreateBuiltinAudioEncoderFactory(),
CreateBuiltinAudioDecoderFactory(), std::move(video_encoder_factory),
diff --git a/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc b/third_party/libwebrtc/pc/peer_connection_end_to_end_unittest.cc
@@ -46,6 +46,7 @@
#include "rtc_base/physical_socket_server.h"
#include "rtc_base/third_party/sigslot/sigslot.h"
#include "rtc_base/thread.h"
+#include "test/create_test_environment.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"
@@ -83,14 +84,15 @@ class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
DataChannelList;
explicit PeerConnectionEndToEndBaseTest(SdpSemantics sdp_semantics)
- : network_thread_(std::make_unique<webrtc::Thread>(&pss_)),
+ : env_(webrtc::CreateTestEnvironment()),
+ network_thread_(std::make_unique<webrtc::Thread>(&pss_)),
worker_thread_(webrtc::Thread::Create()) {
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
caller_ = webrtc::make_ref_counted<PeerConnectionTestWrapper>(
- "caller", &pss_, network_thread_.get(), worker_thread_.get());
+ "caller", env_, &pss_, network_thread_.get(), worker_thread_.get());
callee_ = webrtc::make_ref_counted<PeerConnectionTestWrapper>(
- "callee", &pss_, network_thread_.get(), worker_thread_.get());
+ "callee", env_, &pss_, network_thread_.get(), worker_thread_.get());
webrtc::PeerConnectionInterface::IceServer ice_server;
ice_server.uri = "stun:stun.l.google.com:19302";
config_.servers.push_back(ice_server);
@@ -253,6 +255,7 @@ class PeerConnectionEndToEndBaseTest : public sigslot::has_slots<>,
protected:
webrtc::AutoThread main_thread_;
webrtc::PhysicalSocketServer pss_;
+ webrtc::Environment env_;
std::unique_ptr<webrtc::Thread> network_thread_;
std::unique_ptr<webrtc::Thread> worker_thread_;
webrtc::scoped_refptr<PeerConnectionTestWrapper> caller_;
diff --git a/third_party/libwebrtc/pc/peer_connection_integrationtest.cc b/third_party/libwebrtc/pc/peer_connection_integrationtest.cc
@@ -75,7 +75,6 @@
#include "pc/test/fake_periodic_video_source.h"
#include "pc/test/integration_test_helpers.h"
#include "pc/test/mock_peer_connection_observers.h"
-#include "rtc_base/checks.h"
#include "rtc_base/fake_clock.h"
#include "rtc_base/fake_mdns_responder.h"
#include "rtc_base/fake_network.h"
@@ -90,7 +89,6 @@
#include "rtc_base/string_encode.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/test_certificate_verifier.h"
-#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
#include "system_wrappers/include/metrics.h"
#include "test/gmock.h"
@@ -380,7 +378,7 @@ TEST_P(PeerConnectionIntegrationTest,
FakePeriodicVideoSource::Config config;
config.width = 1280;
config.height = 720;
- config.timestamp_offset_ms = TimeMillis();
+ config.timestamp_offset_ms = env_.clock().TimeInMilliseconds();
caller()->AddTrack(caller()->CreateLocalVideoTrackWithConfig(config));
callee()->AddTrack(callee()->CreateLocalVideoTrackWithConfig(config));
@@ -2524,7 +2522,7 @@ TEST_P(PeerConnectionIntegrationTestWithFakeClock,
// For explanation of these values, see comment above.
static constexpr int required_media_hops = 9;
static constexpr int required_signaling_trips = 2;
- // For internal delays (such as posting an event asychronously).
+ // For internal delays (such as posting an event asynchronously).
static constexpr int allowed_internal_delay_ms = 20;
static constexpr int total_connection_time_ms =
media_hop_delay_ms * required_media_hops +
@@ -2986,7 +2984,7 @@ TEST_P(PeerConnectionIntegrationTest, UnsignaledSsrcGetSourcesVideo) {
// Similar to the above test, except instead of waiting until GetSources() is
// non-empty we wait until media is flowing and then assert that GetSources()
// is not empty. This provides test coverage for https://crbug.com/webrtc/14817
-// where a race due to the re-creationg of the unsignaled ssrc stream would
+// where a race due to the re-creation of the unsignaled ssrc stream would
// clear the GetSources() history. This test not flaking confirms the bug fix.
// TODO(crbug.com/webrtc/441652589): Figure out why this is flaking and
// re-enable the test.
@@ -3632,12 +3630,12 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
- auto start_time_ms = TimeMillis();
+ auto start_time_ms = env_.clock().TimeInMilliseconds();
caller()->CreateAndSetAndSignalOffer();
// We want to stop when the time exceeds one second.
ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
IsRtcOk());
- auto elapsed_time_ms = TimeMillis() - start_time_ms;
+ auto elapsed_time_ms = env_.clock().TimeInMilliseconds() - start_time_ms;
RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
ASSERT_GT(1000, elapsed_time_ms)
<< "Audio transceivers: Negotiation took too long after "
@@ -3670,12 +3668,12 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
- auto start_time_ms = TimeMillis();
+ auto start_time_ms = env_.clock().TimeInMilliseconds();
caller()->CreateAndSetAndSignalOffer();
// We want to stop when the time exceeds one second.
ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
IsRtcOk());
- auto elapsed_time_ms = TimeMillis() - start_time_ms;
+ auto elapsed_time_ms = env_.clock().TimeInMilliseconds() - start_time_ms;
RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
ASSERT_GT(1000, elapsed_time_ms)
<< "Video transceivers: Negotiation took too long after "
@@ -3714,12 +3712,12 @@ TEST_F(PeerConnectionIntegrationTestUnifiedPlan,
}
current_size = caller()->pc()->GetTransceivers().size();
RTC_LOG(LS_INFO) << "Renegotiating with " << current_size << " tracks";
- auto start_time_ms = TimeMillis();
+ auto start_time_ms = env_.clock().TimeInMilliseconds();
caller()->CreateAndSetAndSignalOffer();
// We want to stop when the time exceeds one second.
ASSERT_THAT(WaitUntil([&] { return SignalingStateStable(); }, IsTrue()),
IsRtcOk());
- auto elapsed_time_ms = TimeMillis() - start_time_ms;
+ auto elapsed_time_ms = env_.clock().TimeInMilliseconds() - start_time_ms;
RTC_LOG(LS_INFO) << "Renegotiating took " << elapsed_time_ms << " ms";
// This is a guard against the test using excessive amounts of time.
ASSERT_GT(5000, elapsed_time_ms)
diff --git a/third_party/libwebrtc/pc/rtc_stats_integrationtest.cc b/third_party/libwebrtc/pc/rtc_stats_integrationtest.cc
@@ -21,7 +21,7 @@
#include "api/audio_codecs/builtin_audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/data_channel_interface.h"
-#include "api/field_trials.h"
+#include "api/environment/environment.h"
#include "api/make_ref_counted.h"
#include "api/media_stream_interface.h"
#include "api/media_types.h"
@@ -44,6 +44,7 @@
#include "rtc_base/thread.h"
#include "rtc_base/trace_event.h"
#include "rtc_base/virtual_socket_server.h"
+#include "test/create_test_environment.h"
#include "test/create_test_field_trials.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -62,16 +63,17 @@ constexpr int64_t kGetStatsTimeoutMs = 10000;
class RTCStatsIntegrationTest : public ::testing::Test {
public:
RTCStatsIntegrationTest()
- : network_thread_(new Thread(&virtual_socket_server_)),
+ : env_(CreateTestEnvironment()),
+ network_thread_(new Thread(&virtual_socket_server_)),
worker_thread_(Thread::Create()) {
RTC_CHECK(network_thread_->Start());
RTC_CHECK(worker_thread_->Start());
caller_ = make_ref_counted<PeerConnectionTestWrapper>(
- "caller", &virtual_socket_server_, network_thread_.get(),
+ "caller", env_, &virtual_socket_server_, network_thread_.get(),
worker_thread_.get());
callee_ = make_ref_counted<PeerConnectionTestWrapper>(
- "callee", &virtual_socket_server_, network_thread_.get(),
+ "callee", env_, &virtual_socket_server_, network_thread_.get(),
worker_thread_.get());
}
@@ -159,6 +161,7 @@ class RTCStatsIntegrationTest : public ::testing::Test {
// `network_thread_` uses `virtual_socket_server_` so they must be
// constructed/destructed in the correct order.
+ const Environment env_;
VirtualSocketServer virtual_socket_server_;
std::unique_ptr<Thread> network_thread_;
std::unique_ptr<Thread> worker_thread_;
diff --git a/third_party/libwebrtc/pc/test/integration_test_helpers.cc b/third_party/libwebrtc/pc/test/integration_test_helpers.cc
@@ -21,8 +21,6 @@
#include "api/audio/builtin_audio_processing_builder.h"
#include "api/create_modular_peer_connection_factory.h"
#include "api/enable_media_with_defaults.h"
-#include "api/environment/environment_factory.h"
-#include "api/field_trials_view.h"
#include "api/jsep.h"
#include "api/peer_connection_interface.h"
#include "api/rtc_event_log/rtc_event_log_factory.h"
@@ -223,7 +221,6 @@ bool PeerConnectionIntegrationWrapper::Init(
SocketServer* socket_server,
Thread* network_thread,
Thread* worker_thread,
- std::unique_ptr<FieldTrialsView> field_trials,
std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
@@ -250,7 +247,7 @@ bool PeerConnectionIntegrationWrapper::Init(
pc_factory_dependencies.signaling_thread = signaling_thread;
pc_factory_dependencies.socket_factory = socket_server;
pc_factory_dependencies.network_manager = std::move(network_manager);
- pc_factory_dependencies.env = CreateEnvironment(std::move(field_trials));
+ pc_factory_dependencies.env = env_;
pc_factory_dependencies.decode_metronome =
std::make_unique<TaskQueueMetronome>(TimeDelta::Millis(8));
diff --git a/third_party/libwebrtc/pc/test/integration_test_helpers.h b/third_party/libwebrtc/pc/test/integration_test_helpers.h
@@ -34,7 +34,8 @@
#include "api/crypto/crypto_options.h"
#include "api/data_channel_interface.h"
#include "api/dtls_transport_interface.h"
-#include "api/field_trials_view.h"
+#include "api/environment/environment.h"
+#include "api/environment/environment_factory.h"
#include "api/ice_transport_interface.h"
#include "api/jsep.h"
#include "api/make_ref_counted.h"
@@ -90,9 +91,10 @@
#include "rtc_base/ssl_stream_adapter.h"
#include "rtc_base/task_queue_for_test.h"
#include "rtc_base/thread.h"
-#include "rtc_base/time_utils.h"
#include "rtc_base/virtual_socket_server.h"
+#include "system_wrappers/include/clock.h"
#include "system_wrappers/include/metrics.h"
+#include "test/create_test_environment.h"
#include "test/create_test_field_trials.h"
#include "test/gmock.h"
#include "test/gtest.h"
@@ -362,7 +364,7 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
scoped_refptr<VideoTrackInterface> CreateLocalVideoTrack() {
FakePeriodicVideoSource::Config config;
- config.timestamp_offset_ms = TimeMillis();
+ config.timestamp_offset_ms = env_.clock().TimeInMilliseconds();
return CreateLocalVideoTrackInternal(config);
}
@@ -375,7 +377,7 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
VideoRotation rotation) {
FakePeriodicVideoSource::Config config;
config.rotation = rotation;
- config.timestamp_offset_ms = TimeMillis();
+ config.timestamp_offset_ms = env_.clock().TimeInMilliseconds();
return CreateLocalVideoTrackInternal(config);
}
@@ -793,8 +795,9 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
private:
// Constructor used by friend class PeerConnectionIntegrationBaseTest.
- explicit PeerConnectionIntegrationWrapper(const std::string& debug_name)
- : debug_name_(debug_name) {}
+ explicit PeerConnectionIntegrationWrapper(const std::string& debug_name,
+ Environment env)
+ : debug_name_(debug_name), env_(env) {}
bool Init(const PeerConnectionFactory::Options* options,
const PeerConnectionInterface::RTCConfiguration* config,
@@ -802,7 +805,6 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
SocketServer* socket_server,
Thread* network_thread,
Thread* worker_thread,
- std::unique_ptr<FieldTrialsView> field_trials,
std::unique_ptr<FakeRtcEventLogFactory> event_log_factory,
bool reset_encoder_factory,
bool reset_decoder_factory,
@@ -916,7 +918,6 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
return description;
}
-
// This is a work around to remove unused fake_video_renderers from
// transceivers that have either stopped or are no longer receiving.
void RemoveUnusedVideoRenderers() {
@@ -1145,6 +1146,7 @@ class PeerConnectionIntegrationWrapper : public PeerConnectionObserver,
}
std::string debug_name_;
+ const Environment env_;
// Network manager is owned by the `peer_connection_factory_`.
FakeNetworkManager* fake_network_manager_ = nullptr;
@@ -1378,6 +1380,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
explicit PeerConnectionIntegrationBaseTest(SdpSemantics sdp_semantics)
: sdp_semantics_(sdp_semantics),
+ env_(CreateTestEnvironment()),
ss_(new VirtualSocketServer()),
fss_(new FirewallSocketServer(ss_.get())),
network_thread_(new Thread(fss_.get())),
@@ -1475,17 +1478,20 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
dependencies.cert_generator =
std::make_unique<FakeRTCCertificateGenerator>();
}
- std::unique_ptr<PeerConnectionIntegrationWrapper> client(
- new PeerConnectionIntegrationWrapper(debug_name));
-
std::string field_trials = field_trials_;
auto it = field_trials_overrides_.find(debug_name);
if (it != field_trials_overrides_.end()) {
field_trials = it->second;
}
+ // Override the field trials in the environment.
+ EnvironmentFactory env = EnvironmentFactory(env_);
+ env.Set(CreateTestFieldTrialsPtr(field_trials));
+
+ std::unique_ptr<PeerConnectionIntegrationWrapper> client(
+ new PeerConnectionIntegrationWrapper(debug_name, env.Create()));
+
if (!client->Init(options, &modified_config, std::move(dependencies),
fss_.get(), network_thread_.get(), worker_thread_.get(),
- CreateTestFieldTrialsPtr(field_trials),
std::move(event_log_factory), reset_encoder_factory,
reset_decoder_factory, create_media_engine)) {
return nullptr;
@@ -1722,7 +1728,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
PeerConnectionIntegrationWrapper* caller() { return caller_.get(); }
- // Destroy peerconnections.
+ // Destroy peer connections.
// This can be used to ensure that all pointers to on-stack mocks
// get dropped before exit.
void DestroyPeerConnections() {
@@ -1914,6 +1920,7 @@ class PeerConnectionIntegrationBaseTest : public ::testing::Test {
protected:
SdpSemantics sdp_semantics_;
+ const Environment env_;
private:
AutoThread main_thread_; // Used as the signal thread by most tests.
diff --git a/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.cc b/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.cc
@@ -67,7 +67,7 @@
#include "rtc_base/logging.h"
#include "rtc_base/rtc_certificate_generator.h"
#include "rtc_base/socket_server.h"
-#include "rtc_base/time_utils.h"
+#include "system_wrappers/include/clock.h"
#include "test/gmock.h"
#include "test/gtest.h"
#include "test/wait_until.h"
@@ -154,10 +154,12 @@ void PeerConnectionTestWrapper::AwaitNegotiation(
PeerConnectionTestWrapper::PeerConnectionTestWrapper(
const std::string& name,
+ const webrtc::Environment& env,
webrtc::SocketServer* socket_server,
webrtc::Thread* network_thread,
webrtc::Thread* worker_thread)
: name_(name),
+ env_(env),
socket_server_(socket_server),
network_thread_(network_thread),
worker_thread_(worker_thread),
@@ -186,8 +188,11 @@ bool PeerConnectionTestWrapper::CreatePc(
std::unique_ptr<webrtc::VideoEncoderFactory> video_encoder_factory,
std::unique_ptr<webrtc::VideoDecoderFactory> video_decoder_factory,
std::unique_ptr<webrtc::FieldTrialsView> field_trials) {
+ webrtc::EnvironmentFactory env_factory(env_);
+ env_factory.Set(field_trials.get());
+ Environment env = env_factory.Create();
auto port_allocator = std::make_unique<webrtc::FakePortAllocator>(
- CreateEnvironment(field_trials.get()), socket_server_, network_thread_);
+ env, socket_server_, network_thread_);
RTC_DCHECK_RUN_ON(&pc_thread_checker_);
@@ -554,7 +559,7 @@ PeerConnectionTestWrapper::GetUserMedia(
// Set max frame rate to 10fps to reduce the risk of the tests to be flaky.
webrtc::FakePeriodicVideoSource::Config config;
config.frame_interval_ms = 100;
- config.timestamp_offset_ms = webrtc::TimeMillis();
+ config.timestamp_offset_ms = env_.clock().TimeInMilliseconds();
config.width = resolution.width;
config.height = resolution.height;
diff --git a/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h b/third_party/libwebrtc/pc/test/peer_connection_test_wrapper.h
@@ -20,6 +20,7 @@
#include "api/audio_codecs/audio_encoder_factory.h"
#include "api/audio_options.h"
#include "api/data_channel_interface.h"
+#include "api/environment/environment.h"
#include "api/field_trials_view.h"
#include "api/jsep.h"
#include "api/media_stream_interface.h"
@@ -55,6 +56,7 @@ class PeerConnectionTestWrapper
PeerConnectionTestWrapper* callee);
PeerConnectionTestWrapper(const std::string& name,
+ const webrtc::Environment& env,
webrtc::SocketServer* socket_server,
webrtc::Thread* network_thread,
webrtc::Thread* worker_thread);
@@ -164,6 +166,7 @@ class PeerConnectionTestWrapper
const std::string& candidate);
std::string name_;
+ const webrtc::Environment env_;
webrtc::SocketServer* const socket_server_;
webrtc::Thread* const network_thread_;
webrtc::Thread* const worker_thread_;
