Use RunLoop in call/ tests Bug: webrtc:469327588 Change-Id: Ia88b913a96175f8926208f7fd6bb4d1d6a6a6964 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/461345 Auto-Submit: Evan Shrubsole <eshr@webrtc.org> Commit-Queue: Per Kjellander <perkj@webrtc.org> Reviewed-by: Per Kjellander <perkj@webrtc.org> Cr-Commit-Position: refs/heads/main@{#47312}
diff --git a/call/BUILD.gn b/call/BUILD.gn index a8ad708..815d28a 100644 --- a/call/BUILD.gn +++ b/call/BUILD.gn
@@ -660,6 +660,7 @@ "../test:encoder_settings", "../test:fake_video_codecs", "../test:frame_generator_capturer", + "../test:run_loop", "../test:test_common", "../test:test_flags", "../test:test_support", @@ -699,6 +700,7 @@ "../rtc_base:network_route", "../rtc_base/containers:flat_map", "../rtc_base/network:sent_packet", + "../test:run_loop", "../test:test_support", "//third_party/abseil-cpp/absl/strings:string_view", ] @@ -709,6 +711,7 @@ sources = [ "test/mock_bitrate_allocator.h" ] deps = [ ":bitrate_allocator", + "../test:run_loop", "../test:test_support", ] } @@ -730,6 +733,7 @@ "../api/audio_codecs:audio_codecs_api", "../api/crypto:frame_decryptor_interface", "../api/transport/rtp:rtp_source", + "../test:run_loop", "../test:test_support", ] } @@ -760,6 +764,7 @@ "../rtc_base:checks", "../rtc_base:copy_on_write_buffer", "../system_wrappers", + "../test:run_loop", "../test:test_support", "../test/network:simulated_network", ]
diff --git a/call/adaptation/BUILD.gn b/call/adaptation/BUILD.gn index 3149c13..0b0da34 100644 --- a/call/adaptation/BUILD.gn +++ b/call/adaptation/BUILD.gn
@@ -89,6 +89,7 @@ "../../rtc_base:task_queue_for_test", "../../rtc_base:threading", "../../test:create_test_field_trials", + "../../test:run_loop", "../../test:test_support", "../../test:wait_until", "../../video/config:encoder_config",
diff --git a/call/adaptation/resource_adaptation_processor_unittest.cc b/call/adaptation/resource_adaptation_processor_unittest.cc index 4f63960..413f6fe 100644 --- a/call/adaptation/resource_adaptation_processor_unittest.cc +++ b/call/adaptation/resource_adaptation_processor_unittest.cc
@@ -27,11 +27,11 @@ #include "call/adaptation/video_stream_input_state_provider.h" #include "rtc_base/event.h" #include "rtc_base/task_queue_for_test.h" -#include "rtc_base/thread.h" #include "rtc_base/thread_annotations.h" #include "test/create_test_field_trials.h" #include "test/gmock.h" #include "test/gtest.h" +#include "test/run_loop.h" #include "test/wait_until.h" namespace webrtc { @@ -141,12 +141,10 @@ processor_.reset(); } - static void WaitUntilTaskQueueIdle() { - ASSERT_TRUE(Thread::Current()->ProcessMessages(0)); - } + void WaitUntilTaskQueueIdle() { main_thread_.Flush(); } protected: - AutoThread main_thread_; + test::RunLoop main_thread_; FakeFrameRateProvider frame_rate_provider_; VideoStreamInputStateProvider input_state_provider_; scoped_refptr<FakeResource> resource_;
diff --git a/call/flexfec_receive_stream_unittest.cc b/call/flexfec_receive_stream_unittest.cc index 27c8d2c..c4df175 100644 --- a/call/flexfec_receive_stream_unittest.cc +++ b/call/flexfec_receive_stream_unittest.cc
@@ -25,10 +25,10 @@ #include "modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" #include "modules/rtp_rtcp/source/byte_io.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h" -#include "rtc_base/thread.h" #include "test/gmock.h" #include "test/gtest.h" #include "test/mock_transport.h" +#include "test/run_loop.h" namespace webrtc { @@ -94,7 +94,7 @@ receive_stream_->UnregisterFromTransport(); } - AutoThread main_thread_; + test::RunLoop main_thread_; MockTransport rtcp_send_transport_; MockRtcEventLog log_; FlexfecReceiveStream::Config config_;
diff --git a/call/rtp_transport_controller_send_unittest.cc b/call/rtp_transport_controller_send_unittest.cc index 3e07614..5bf29ea7 100644 --- a/call/rtp_transport_controller_send_unittest.cc +++ b/call/rtp_transport_controller_send_unittest.cc
@@ -23,10 +23,10 @@ #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/source/rtcp_packet/congestion_control_feedback.h" #include "rtc_base/containers/flat_map.h" -#include "rtc_base/thread.h" #include "test/create_test_environment.h" #include "test/gmock.h" #include "test/gtest.h" +#include "test/run_loop.h" namespace webrtc { namespace { @@ -102,7 +102,7 @@ TEST(RtpTransportControllerSendTest, IgnoresFeedbackForReportedReceivedPacketThatWereNotSent) { - AutoThread main_thread; + test::RunLoop main_thread; RtpTransportControllerSend transport({.env = CreateTestEnvironment()}); transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB); PacketSender sender(transport); @@ -131,7 +131,7 @@ AccumulatesNumberOfReportedReceivedPacketsPerSsrcPerEcnMarkingType) { constexpr uint32_t kSsrc1 = 1'000; constexpr uint32_t kSsrc2 = 2'000; - AutoThread main_thread; + test::RunLoop main_thread; RtpTransportControllerSend transport({.env = CreateTestEnvironment()}); transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB); @@ -176,7 +176,7 @@ } TEST(RtpTransportControllerSendTest, CalculatesNumberOfBleachedPackets) { - AutoThread main_thread; + test::RunLoop main_thread; RtpTransportControllerSend transport({.env = CreateTestEnvironment()}); transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB); PacketSender sender(transport); @@ -211,7 +211,7 @@ TEST(RtpTransportControllerSendTest, AccumulatesNumberOfReportedLostAndRecoveredPackets) { - AutoThread main_thread; + test::RunLoop main_thread; RtpTransportControllerSend transport({.env = CreateTestEnvironment()}); transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB); @@ -258,7 +258,7 @@ TEST(RtpTransportControllerSendTest, DoesNotCountGapsInSequenceNumberBetweenReportsAsLoss) { - AutoThread main_thread; + test::RunLoop main_thread; RtpTransportControllerSend transport({.env = CreateTestEnvironment()}); transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);