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Replace gunit.h macros with WaitUntil in audio/

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Bug: webrtc:381524905
Change-Id: If153ae0f284873a35cfce7d72e3dcba41f768190
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/373741
Commit-Queue: Per Åhgren <peah@webrtc.org>
Auto-Submit: Evan Shrubsole <eshr@webrtc.org>
Reviewed-by: Per Åhgren <peah@webrtc.org>
Cr-Commit-Position: refs/heads/main@{#43717}
This commit is contained in:
Evan Shrubsole 2025-01-07 14:04:29 +00:00 committed by WebRTC LUCI CQ
parent f3dafd1d43
commit 815c5c0179
3 changed files with 36 additions and 11 deletions
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5
audio/BUILD.gn
View File

@ -177,9 +177,11 @@ if (rtc_include_tests) {
"../api:mock_frame_encryptor", "../api:mock_frame_encryptor",
"../api:mock_frame_transformer", "../api:mock_frame_transformer",
"../api:mock_transformable_audio_frame", "../api:mock_transformable_audio_frame",
"../api:rtc_error_matchers",
"../api:rtp_headers", "../api:rtp_headers",
"../api:rtp_parameters", "../api:rtp_parameters",
"../api:scoped_refptr", "../api:scoped_refptr",
"../api:simulated_network_api",
"../api:transport_api", "../api:transport_api",
"../api/audio:audio_frame_api", "../api/audio:audio_frame_api",
"../api/audio:audio_mixer_api", "../api/audio:audio_mixer_api",
@ -193,6 +195,7 @@ if (rtc_include_tests) {
"../api/crypto:options", "../api/crypto:options",
"../api/environment", "../api/environment",
"../api/environment:environment_factory", "../api/environment:environment_factory",
"../api/task_queue",
"../api/task_queue:default_task_queue_factory", "../api/task_queue:default_task_queue_factory",
"../api/task_queue/test:mock_task_queue_base", "../api/task_queue/test:mock_task_queue_base",
"../api/transport:bitrate_settings", "../api/transport:bitrate_settings",
@ -240,8 +243,10 @@ if (rtc_include_tests) {
"../test:test_common", "../test:test_common",
"../test:test_support", "../test:test_support",
"../test:video_test_constants", "../test:video_test_constants",
"../test:wait_until",
"../test/time_controller:time_controller", "../test/time_controller:time_controller",
"utility:utility_tests", "utility:utility_tests",
"//testing/gmock",
"//testing/gtest", "//testing/gtest",
"//third_party/abseil-cpp/absl/memory", "//third_party/abseil-cpp/absl/memory",
"//third_party/abseil-cpp/absl/strings:string_view", "//third_party/abseil-cpp/absl/strings:string_view",

22
audio/channel_send_unittest.cc
View File

@ -34,6 +34,7 @@
#include "api/scoped_refptr.h" #include "api/scoped_refptr.h"
#include "api/test/mock_frame_transformer.h" #include "api/test/mock_frame_transformer.h"
#include "api/test/mock_transformable_audio_frame.h" #include "api/test/mock_transformable_audio_frame.h"
#include "api/test/rtc_error_matchers.h"
#include "api/transport/bitrate_settings.h" #include "api/transport/bitrate_settings.h"
#include "api/units/data_rate.h" #include "api/units/data_rate.h"
#include "api/units/time_delta.h" #include "api/units/time_delta.h"
@ -42,18 +43,20 @@
#include "call/rtp_transport_controller_send.h" #include "call/rtp_transport_controller_send.h"
#include "modules/rtp_rtcp/include/rtp_rtcp_defines.h" #include "modules/rtp_rtcp/include/rtp_rtcp_defines.h"
#include "modules/rtp_rtcp/source/rtp_packet_received.h" #include "modules/rtp_rtcp/source/rtp_packet_received.h"
#include "rtc_base/gunit.h"
#include "test/gmock.h" #include "test/gmock.h"
#include "test/gtest.h" #include "test/gtest.h"
#include "test/mock_transport.h" #include "test/mock_transport.h"
#include "test/scoped_key_value_config.h" #include "test/scoped_key_value_config.h"
#include "test/time_controller/simulated_time_controller.h" #include "test/time_controller/simulated_time_controller.h"
#include "test/wait_until.h"
namespace webrtc { namespace webrtc {
namespace voe { namespace voe {
namespace { namespace {
using ::testing::Eq;
using ::testing::Invoke; using ::testing::Invoke;
using ::testing::IsTrue;
using ::testing::NiceMock; using ::testing::NiceMock;
using ::testing::Return; using ::testing::Return;
using ::testing::SaveArg; using ::testing::SaveArg;
@ -212,9 +215,11 @@ TEST_F(ChannelSendTest, FrameTransformerGetsCorrectTimestamp) {
// Ensure the RTP timestamp on the frame passed to the transformer // Ensure the RTP timestamp on the frame passed to the transformer
// includes the RTP offset and matches the actual RTP timestamp on the sent // includes the RTP offset and matches the actual RTP timestamp on the sent
// packet. // packet.
EXPECT_EQ_WAIT(transformable_frame_timestamp, EXPECT_THAT(
0 + channel_->GetRtpRtcp()->StartTimestamp(), 1000); WaitUntil([&] { return 0 + channel_->GetRtpRtcp()->StartTimestamp(); },
EXPECT_TRUE_WAIT(sent_timestamp, 1000); Eq(transformable_frame_timestamp)),
IsRtcOk());
EXPECT_THAT(WaitUntil([&] { return sent_timestamp; }, IsTrue()), IsRtcOk());
EXPECT_EQ(*sent_timestamp, transformable_frame_timestamp); EXPECT_EQ(*sent_timestamp, transformable_frame_timestamp);
} }
@ -264,7 +269,7 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToCorrectTransformedFrame) {
ProcessNextFrame(CreateAudioFrame(/*data_init_value=*/3)); ProcessNextFrame(CreateAudioFrame(/*data_init_value=*/3));
// Wait for both packets to be encoded and sent to the transform. // Wait for both packets to be encoded and sent to the transform.
EXPECT_EQ_WAIT(frames.size(), 2ul, 1000); EXPECT_THAT(WaitUntil([&] { return frames.size(); }, Eq(2ul)), IsRtcOk());
// Complete the transforms on both frames at the same time // Complete the transforms on both frames at the same time
callback->OnTransformedFrame(std::move(frames[0])); callback->OnTransformedFrame(std::move(frames[0]));
callback->OnTransformedFrame(std::move(frames[1])); callback->OnTransformedFrame(std::move(frames[1]));
@ -274,7 +279,8 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToCorrectTransformedFrame) {
// Ensure the audio levels on both sent packets is present and // Ensure the audio levels on both sent packets is present and
// matches their contents. // matches their contents.
EXPECT_EQ_WAIT(sent_audio_levels.size(), 2ul, 1000); EXPECT_THAT(WaitUntil([&] { return sent_audio_levels.size(); }, Eq(2ul)),
IsRtcOk());
// rms dbov of the packet with raw audio of 7s is 73. // rms dbov of the packet with raw audio of 7s is 73.
EXPECT_EQ(sent_audio_levels[0], 73); EXPECT_EQ(sent_audio_levels[0], 73);
// rms dbov of the second packet with raw audio of 3s is 81. // rms dbov of the second packet with raw audio of 3s is 81.
@ -318,14 +324,14 @@ TEST_F(ChannelSendTest, AudioLevelsAttachedToInsertedTransformedFrame) {
uint8_t payload[10]; uint8_t payload[10];
ON_CALL(*mock_frame, GetData()) ON_CALL(*mock_frame, GetData())
.WillByDefault(Return(rtc::ArrayView<uint8_t>(&payload[0], 10))); .WillByDefault(Return(rtc::ArrayView<uint8_t>(&payload[0], 10)));
EXPECT_TRUE_WAIT(callback, 1000); EXPECT_THAT(WaitUntil([&] { return callback; }, IsTrue()), IsRtcOk());
callback->OnTransformedFrame(std::move(mock_frame)); callback->OnTransformedFrame(std::move(mock_frame));
// Allow things posted back to the encoder queue to run. // Allow things posted back to the encoder queue to run.
time_controller_.AdvanceTime(TimeDelta::Millis(10)); time_controller_.AdvanceTime(TimeDelta::Millis(10));
// Ensure the audio levels is set on the sent packet. // Ensure the audio levels is set on the sent packet.
EXPECT_TRUE_WAIT(sent_audio_level, 1000); EXPECT_THAT(WaitUntil([&] { return sent_audio_level; }, IsTrue()), IsRtcOk());
EXPECT_EQ(*sent_audio_level, audio_level); EXPECT_EQ(*sent_audio_level, audio_level);
} }

20
audio/test/non_sender_rtt_test.cc
View File

@ -8,15 +8,26 @@
* be found in the AUTHORS file in the root of the source tree. * be found in the AUTHORS file in the root of the source tree.
*/ */
#include <cstdint>
#include <vector>
#include "api/task_queue/task_queue_base.h"
#include "api/test/rtc_error_matchers.h"
#include "api/test/simulated_network.h"
#include "api/units/time_delta.h"
#include "audio/test/audio_end_to_end_test.h" #include "audio/test/audio_end_to_end_test.h"
#include "rtc_base/gunit.h" #include "call/audio_receive_stream.h"
#include "call/audio_send_stream.h"
#include "rtc_base/task_queue_for_test.h" #include "rtc_base/task_queue_for_test.h"
#include "system_wrappers/include/sleep.h" #include "test/call_test.h"
#include "test/gmock.h"
#include "test/gtest.h" #include "test/gtest.h"
#include "test/wait_until.h"
namespace webrtc { namespace webrtc {
namespace test { namespace test {
using ::testing::IsTrue;
using NonSenderRttTest = CallTest; using NonSenderRttTest = CallTest;
TEST_F(NonSenderRttTest, NonSenderRttStats) { TEST_F(NonSenderRttTest, NonSenderRttStats) {
@ -47,7 +58,10 @@ TEST_F(NonSenderRttTest, NonSenderRttStats) {
// Wait until we have an RTT measurement, but no longer than // Wait until we have an RTT measurement, but no longer than
// `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare // `kLongTimeoutMs`. This usually takes around 5 seconds, but in rare
// cases it can take more than 10 seconds. // cases it can take more than 10 seconds.
EXPECT_TRUE_WAIT(HasRoundTripTimeMeasurement(), kLongTimeoutMs); EXPECT_THAT(
WaitUntil([&] { return HasRoundTripTimeMeasurement(); }, IsTrue(),
{.timeout = webrtc::TimeDelta::Millis(kLongTimeoutMs)}),
IsRtcOk());
} }
void OnStreamsStopped() override { void OnStreamsStopped() override {