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}
This commit is contained in:
Evan Shrubsole 2026-03-31 08:16:39 +00:00 committed by WebRTC LUCI CQ
parent 06d113a66c
commit 2dab14a4fc
5 changed files with 17 additions and 13 deletions

View File

@ -660,6 +660,7 @@ if (rtc_include_tests) {
"../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 @@ if (rtc_include_tests) {
"../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 @@ if (rtc_include_tests) {
sources = [ "test/mock_bitrate_allocator.h" ]
deps = [
":bitrate_allocator",
"../test:run_loop",
"../test:test_support",
]
}
@ -730,6 +733,7 @@ if (rtc_include_tests) {
"../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 @@ if (rtc_include_tests) {
"../rtc_base:checks",
"../rtc_base:copy_on_write_buffer",
"../system_wrappers",
"../test:run_loop",
"../test:test_support",
"../test/network:simulated_network",
]

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@ -89,6 +89,7 @@ if (rtc_include_tests) {
"../../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",

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@ -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 @@ class ResourceAdaptationProcessorTest : public ::testing::Test {
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_;

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@ -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 @@ class FlexfecReceiveStreamTest : public ::testing::Test {
receive_stream_->UnregisterFromTransport();
}
AutoThread main_thread_;
test::RunLoop main_thread_;
MockTransport rtcp_send_transport_;
MockRtcEventLog log_;
FlexfecReceiveStream::Config config_;

View File

@ -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 @@ rtcp::CongestionControlFeedback GenerateFeedback(
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 @@ TEST(RtpTransportControllerSendTest,
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,
}
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, CalculatesNumberOfBleachedPackets) {
TEST(RtpTransportControllerSendTest,
AccumulatesNumberOfReportedLostAndRecoveredPackets) {
AutoThread main_thread;
test::RunLoop main_thread;
RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);
@ -258,7 +258,7 @@ TEST(RtpTransportControllerSendTest,
TEST(RtpTransportControllerSendTest,
DoesNotCountGapsInSequenceNumberBetweenReportsAsLoss) {
AutoThread main_thread;
test::RunLoop main_thread;
RtpTransportControllerSend transport({.env = CreateTestEnvironment()});
transport.SetPreferredRtcpCcAckType(RtcpFeedbackType::CCFB);