diff --git a/audio/BUILD.gn b/audio/BUILD.gn index 066cce22c6..999851df39 100644 --- a/audio/BUILD.gn +++ b/audio/BUILD.gn @@ -162,6 +162,7 @@ if (rtc_include_tests) { "../rtc_base:rtc_base_approved", "../rtc_base:rtc_task_queue", "../rtc_base:safe_compare", + "../rtc_base:timeutils", "../system_wrappers:system_wrappers", "../test:audio_codec_mocks", "../test:rtp_test_utils", diff --git a/audio/audio_receive_stream_unittest.cc b/audio/audio_receive_stream_unittest.cc index cc1e9d4513..b64626f55a 100644 --- a/audio/audio_receive_stream_unittest.cc +++ b/audio/audio_receive_stream_unittest.cc @@ -25,6 +25,7 @@ #include "modules/bitrate_controller/include/mock/mock_bitrate_controller.h" #include "modules/pacing/packet_router.h" #include "modules/rtp_rtcp/source/byte_io.h" +#include "rtc_base/time_utils.h" #include "test/gtest.h" #include "test/mock_audio_decoder_factory.h" #include "test/mock_transport.h" diff --git a/logging/rtc_event_log/rtc_event_log_impl.cc b/logging/rtc_event_log/rtc_event_log_impl.cc index 66a7fb7ee2..0dbda0f2da 100644 --- a/logging/rtc_event_log/rtc_event_log_impl.cc +++ b/logging/rtc_event_log/rtc_event_log_impl.cc @@ -31,6 +31,7 @@ #include "rtc_base/sequenced_task_checker.h" #include "rtc_base/task_queue.h" #include "rtc_base/thread_annotations.h" +#include "rtc_base/time_utils.h" namespace webrtc { diff --git a/modules/audio_coding/BUILD.gn b/modules/audio_coding/BUILD.gn index 1f034ebc2d..074ca93d78 100644 --- a/modules/audio_coding/BUILD.gn +++ b/modules/audio_coding/BUILD.gn @@ -2042,6 +2042,7 @@ if (rtc_include_tests) { "../../rtc_base:rtc_base_approved", "../../rtc_base:rtc_base_tests_utils", "../../rtc_base:sanitizer", + "../../rtc_base:timeutils", "../../rtc_base/system:arch", "../../system_wrappers", "../../system_wrappers:cpu_features_api", diff --git a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc index 42189c3e11..b1e3313289 100644 --- a/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc +++ b/modules/audio_coding/audio_network_adaptor/event_log_writer_unittest.cc @@ -13,6 +13,7 @@ #include "logging/rtc_event_log/events/rtc_event_audio_network_adaptation.h" #include "logging/rtc_event_log/mock/mock_rtc_event_log.h" #include "modules/audio_coding/audio_network_adaptor/event_log_writer.h" +#include "rtc_base/checks.h" #include "test/gtest.h" namespace webrtc { diff --git a/test/call_test.h b/test/call_test.h index c0ba1ff2a3..fb2605198f 100644 --- a/test/call_test.h +++ b/test/call_test.h @@ -10,7 +10,9 @@ #ifndef TEST_CALL_TEST_H_ #define TEST_CALL_TEST_H_ +#include #include +#include #include #include "api/test/video/function_video_decoder_factory.h" diff --git a/test/scenario/call_client.h b/test/scenario/call_client.h index 793e16d4a6..09c1545a91 100644 --- a/test/scenario/call_client.h +++ b/test/scenario/call_client.h @@ -9,8 +9,11 @@ */ #ifndef TEST_SCENARIO_CALL_CLIENT_H_ #define TEST_SCENARIO_CALL_CLIENT_H_ + +#include #include #include +#include #include #include "call/call.h" diff --git a/video/video_analyzer.h b/video/video_analyzer.h index fffc784cf2..8226270730 100644 --- a/video/video_analyzer.h +++ b/video/video_analyzer.h @@ -17,6 +17,7 @@ #include #include "api/video/video_source_interface.h" +#include "rtc_base/time_utils.h" #include "test/layer_filtering_transport.h" #include "test/rtp_file_writer.h" #include "test/statistics.h"