Enabling `gn check` on webrtc/test
BUG=webrtc:6828
NOTRY=True
Review-Url: https://codereview.webrtc.org/2911203002
Cr-Commit-Position: refs/heads/master@{#18372}
diff --git a/.gn b/.gn
index a6b4af7..70a0c3e 100644
--- a/.gn
+++ b/.gn
@@ -37,6 +37,7 @@
"//webrtc/sdk/*",
"//webrtc/stats/*",
"//webrtc/system_wrappers/*",
+ "//webrtc/test/*",
"//webrtc/tools/*",
"//webrtc/video/*",
"//webrtc/voice_engine/*",
diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn
index 1b62973..8f87576 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -1103,6 +1103,118 @@
]
}
+rtc_source_set("neteq_test_tools") {
+ testonly = true
+ sources = [
+ "neteq/tools/audio_checksum.h",
+ "neteq/tools/audio_loop.cc",
+ "neteq/tools/audio_loop.h",
+ "neteq/tools/constant_pcm_packet_source.cc",
+ "neteq/tools/constant_pcm_packet_source.h",
+ "neteq/tools/output_audio_file.h",
+ "neteq/tools/output_wav_file.h",
+ "neteq/tools/rtp_file_source.cc",
+ "neteq/tools/rtp_file_source.h",
+ "neteq/tools/rtp_generator.cc",
+ "neteq/tools/rtp_generator.h",
+ ]
+
+ public_configs = [ ":neteq_tools_config" ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ deps = [
+ ":pcm16b",
+ "..:module_api",
+ "../..:webrtc_common",
+ "../../base:rtc_base_approved",
+ "../../base:rtc_base_tests_utils",
+ "../../common_audio",
+ "../../test:rtp_test_utils",
+ "../rtp_rtcp",
+ ]
+
+ public_deps = [
+ ":neteq_tools",
+ ":neteq_tools_minimal",
+ ]
+
+ if (rtc_enable_protobuf) {
+ sources += [
+ "neteq/tools/neteq_packet_source_input.cc",
+ "neteq/tools/neteq_packet_source_input.h",
+ ]
+ deps += [ ":rtc_event_log_source" ]
+ }
+}
+
+config("neteq_tools_config") {
+ include_dirs = [ "tools" ]
+}
+
+rtc_source_set("neteq_tools") {
+ sources = [
+ "neteq/tools/fake_decode_from_file.cc",
+ "neteq/tools/fake_decode_from_file.h",
+ "neteq/tools/input_audio_file.cc",
+ "neteq/tools/input_audio_file.h",
+ "neteq/tools/neteq_replacement_input.cc",
+ "neteq/tools/neteq_replacement_input.h",
+ "neteq/tools/resample_input_audio_file.cc",
+ "neteq/tools/resample_input_audio_file.h",
+ ]
+
+ public_configs = [ ":neteq_tools_config" ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ deps = [
+ "../..:webrtc_common",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../base:rtc_base_approved",
+ "../../common_audio",
+ "../rtp_rtcp",
+ ]
+
+ public_deps = [
+ ":neteq_tools_minimal",
+ ]
+}
+
+if (rtc_enable_protobuf) {
+ rtc_static_library("rtc_event_log_source") {
+ testonly = true
+
+ # TODO(kjellander): Remove (bugs.webrtc.org/6828)
+ # Needs call.h to be moved to webrtc/api first.
+ check_includes = false
+
+ sources = [
+ "neteq/tools/rtc_event_log_source.cc",
+ "neteq/tools/rtc_event_log_source.h",
+ ]
+
+ if (!build_with_chromium && is_clang) {
+ # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
+ suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
+ }
+
+ deps = [
+ "../../base:rtc_base_approved",
+ "../../logging:rtc_event_log_parser",
+ ]
+ public_deps = [
+ "../../logging:rtc_event_log_proto",
+ ]
+ }
+}
+
if (rtc_include_tests) {
group("audio_coding_tests") {
testonly = true
@@ -1395,32 +1507,6 @@
proto_out_dir = "webrtc/modules/audio_coding/neteq"
}
- rtc_static_library("rtc_event_log_source") {
- testonly = true
-
- # TODO(kjellander): Remove (bugs.webrtc.org/6828)
- # Needs call.h to be moved to webrtc/api first.
- check_includes = false
-
- sources = [
- "neteq/tools/rtc_event_log_source.cc",
- "neteq/tools/rtc_event_log_source.h",
- ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-
- deps = [
- "../../base:rtc_base_approved",
- "../../logging:rtc_event_log_parser",
- ]
- public_deps = [
- "../../logging:rtc_event_log_proto",
- ]
- }
-
rtc_test("neteq_rtpplay") {
testonly = true
defines = []
@@ -1542,90 +1628,6 @@
]
}
- config("neteq_tools_config") {
- include_dirs = [ "tools" ]
- }
-
- rtc_source_set("neteq_tools") {
- sources = [
- "neteq/tools/fake_decode_from_file.cc",
- "neteq/tools/fake_decode_from_file.h",
- "neteq/tools/input_audio_file.cc",
- "neteq/tools/input_audio_file.h",
- "neteq/tools/neteq_replacement_input.cc",
- "neteq/tools/neteq_replacement_input.h",
- "neteq/tools/resample_input_audio_file.cc",
- "neteq/tools/resample_input_audio_file.h",
- ]
-
- public_configs = [ ":neteq_tools_config" ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-
- deps = [
- "../..:webrtc_common",
- "../../api/audio_codecs:audio_codecs_api",
- "../../base:rtc_base_approved",
- "../../common_audio",
- "../rtp_rtcp",
- ]
-
- public_deps = [
- ":neteq_tools_minimal",
- ]
- }
-
- rtc_source_set("neteq_test_tools") {
- testonly = true
- sources = [
- "neteq/tools/audio_checksum.h",
- "neteq/tools/audio_loop.cc",
- "neteq/tools/audio_loop.h",
- "neteq/tools/constant_pcm_packet_source.cc",
- "neteq/tools/constant_pcm_packet_source.h",
- "neteq/tools/output_audio_file.h",
- "neteq/tools/output_wav_file.h",
- "neteq/tools/rtp_file_source.cc",
- "neteq/tools/rtp_file_source.h",
- "neteq/tools/rtp_generator.cc",
- "neteq/tools/rtp_generator.h",
- ]
-
- public_configs = [ ":neteq_tools_config" ]
-
- if (!build_with_chromium && is_clang) {
- # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
- suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
- }
-
- deps = [
- ":pcm16b",
- "..:module_api",
- "../..:webrtc_common",
- "../../base:rtc_base_approved",
- "../../base:rtc_base_tests_utils",
- "../../common_audio",
- "../../test:rtp_test_utils",
- "../rtp_rtcp",
- ]
-
- public_deps = [
- ":neteq_tools",
- ":neteq_tools_minimal",
- ]
-
- if (rtc_enable_protobuf) {
- sources += [
- "neteq/tools/neteq_packet_source_input.cc",
- "neteq/tools/neteq_packet_source_input.h",
- ]
- deps += [ ":rtc_event_log_source" ]
- }
- }
-
rtc_source_set("neteq_test_tools_deprecated") {
testonly = true
sources = [
diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn
index 2f678d6..6476831 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -254,6 +254,19 @@
}
}
+rtc_source_set("mock_audio_device") {
+ testonly = true
+ sources = [
+ "include/mock_audio_device.h",
+ "include/mock_audio_transport.h",
+ ]
+ deps = [
+ ":audio_device",
+ "../../test:test_support",
+ ]
+ all_dependent_configs = [ ":mock_audio_device_config" ]
+}
+
if (rtc_include_tests) {
rtc_source_set("audio_device_unittests") {
testonly = true
@@ -309,19 +322,6 @@
}
}
- rtc_source_set("mock_audio_device") {
- testonly = true
- sources = [
- "include/mock_audio_device.h",
- "include/mock_audio_transport.h",
- ]
- deps = [
- ":audio_device",
- "../../test:test_support",
- ]
- all_dependent_configs = [ ":mock_audio_device_config" ]
- }
-
if (!is_ios) {
# These tests do not work on ios, see
# https://bugs.chromium.org/p/webrtc/issues/detail?id=4755
diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn
index 7f2d64f..a9e257e 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -211,6 +211,21 @@
}
}
+rtc_source_set("mock_rtp_rtcp") {
+ testonly = true
+ sources = [
+ "mocks/mock_recovered_packet_receiver.h",
+ "mocks/mock_rtcp_rtt_stats.h",
+ "mocks/mock_rtp_rtcp.h",
+ ]
+ deps = [
+ ":rtp_rtcp",
+ "..:module_api",
+ "../../base:rtc_base_approved",
+ "../../test:test_support",
+ ]
+}
+
if (rtc_include_tests) {
rtc_executable("test_packet_masks_metrics") {
testonly = true
@@ -250,21 +265,6 @@
}
}
- rtc_source_set("mock_rtp_rtcp") {
- testonly = true
- sources = [
- "mocks/mock_recovered_packet_receiver.h",
- "mocks/mock_rtcp_rtt_stats.h",
- "mocks/mock_rtp_rtcp.h",
- ]
- deps = [
- ":rtp_rtcp",
- "..:module_api",
- "../../base:rtc_base_approved",
- "../../test:test_support",
- ]
- }
-
rtc_source_set("rtp_rtcp_unittests") {
testonly = true
diff --git a/webrtc/test/BUILD.gn b/webrtc/test/BUILD.gn
index bb04e19..9b42b25 100644
--- a/webrtc/test/BUILD.gn
+++ b/webrtc/test/BUILD.gn
@@ -58,9 +58,14 @@
}
deps = [
+ "..:video_stream_api",
+ "..:webrtc_common",
+ "../base:rtc_base_approved",
+ "../base:rtc_task_queue",
"../common_video",
"../media:rtc_media_base",
"../modules/video_capture:video_capture_module",
+ "../system_wrappers",
]
}
@@ -82,6 +87,7 @@
deps = [
"..:webrtc_common",
+ "../base:rtc_base_approved",
"../modules/rtp_rtcp",
"//testing/gtest",
]
@@ -165,6 +171,7 @@
]
deps = [
":field_trial",
+ "../base:rtc_base_approved",
"../system_wrappers:metrics_default",
"//testing/gmock",
"//testing/gtest",
@@ -188,6 +195,9 @@
]
deps = [
+ ":test_support",
+ ":video_test_common",
+ "..:webrtc_common",
"../base:rtc_base_approved",
"../common_video",
"../system_wrappers",
@@ -252,7 +262,16 @@
}
rtc_test("test_support_unittests") {
- deps = []
+ deps = [
+ ":fake_audio_device",
+ ":rtp_test_utils",
+ "../api:video_frame_api",
+ "../base:rtc_base_approved",
+ "../call:call_interfaces",
+ "../common_audio",
+ "../modules/rtp_rtcp",
+ "../system_wrappers",
+ ]
sources = [
"fake_audio_device_unittest.cc",
"fake_network_pipe_unittest.cc",
@@ -314,11 +333,16 @@
"testsupport/fileutils.cc",
"testsupport/fileutils.h",
]
+ deps = [
+ "..:webrtc_common",
+ "../base:rtc_base_approved",
+ ]
if (is_ios) {
sources += [ "testsupport/iosfileutils.mm" ]
- deps = [
- "../sdk:objc_common",
- ]
+ deps += [ "../sdk:objc_common" ]
+ }
+ if (is_win) {
+ deps += [ "../base:rtc_base" ]
}
visibility = [ ":*" ]
}
@@ -343,6 +367,7 @@
]
deps = [
":fileutils",
+ ":test_support",
"//testing/gmock",
"//testing/gtest",
]
@@ -361,9 +386,12 @@
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
+ "..:webrtc_common",
"../api:transport_api",
"../base:rtc_base_approved",
"../call",
+ "../modules/rtp_rtcp",
+ "../system_wrappers",
]
}
@@ -378,8 +406,11 @@
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
deps = [
+ "..:webrtc_common",
"../base:rtc_base_approved",
+ "../common_audio:common_audio",
"../modules/audio_device:audio_device",
+ "../system_wrappers:system_wrappers",
]
}
@@ -431,15 +462,30 @@
":rtp_test_utils",
":test_support",
":video_test_common",
+ "..:video_stream_api",
"..:webrtc_common",
+ "../api:transport_api",
+ "../api:video_frame_api",
+ "../api/audio_codecs:builtin_audio_decoder_factory",
"../api/audio_codecs:builtin_audio_encoder_factory",
"../api/video_codecs:video_codecs_api",
"../audio",
"../base:rtc_base_approved",
+ "../base:rtc_task_queue",
"../call",
+ "../common_video",
+ "../logging:rtc_event_log_api",
+ "../modules/audio_device:mock_audio_device",
"../modules/audio_mixer:audio_mixer_impl",
"../modules/audio_processing",
+ "../modules/rtp_rtcp",
+ "../modules/rtp_rtcp:mock_rtp_rtcp",
+ "../modules/video_coding:webrtc_h264",
+ "../modules/video_coding:webrtc_vp8",
+ "../modules/video_coding:webrtc_vp9",
+ "../system_wrappers",
"../video",
+ "../voice_engine",
"//testing/gmock",
"//testing/gtest",
]
@@ -515,6 +561,9 @@
deps = [
":test_support",
+ "..:webrtc_common",
+ "../base:rtc_base_approved",
+ "../common_video",
"../modules/media_file",
"//testing/gtest",
]
@@ -531,7 +580,10 @@
]
deps = [
+ ":test_support",
"../api/audio_codecs:audio_codecs_api",
+ "../api/audio_codecs:builtin_audio_decoder_factory",
+ "../base:rtc_base_approved",
"//testing/gmock",
]
}
diff --git a/webrtc/test/fuzzers/BUILD.gn b/webrtc/test/fuzzers/BUILD.gn
index 33c2c47..6e5098e 100644
--- a/webrtc/test/fuzzers/BUILD.gn
+++ b/webrtc/test/fuzzers/BUILD.gn
@@ -15,6 +15,7 @@
"webrtc_fuzzer_main.cc",
]
deps = [
+ "../../base:rtc_base_approved",
"../../system_wrappers:field_trial_default",
"../../system_wrappers:metrics_default",
"//testing/libfuzzer:libfuzzer_main",
@@ -64,6 +65,7 @@
"vp8_qp_parser_fuzzer.cc",
]
deps = [
+ "../../modules/video_coding:video_coding_utility",
"../../modules/video_coding/",
]
}
@@ -73,6 +75,7 @@
"h264_bitstream_parser_fuzzer.cc",
]
deps = [
+ "../../common_video",
"../../modules/video_coding/",
]
}
@@ -82,6 +85,7 @@
"flexfec_header_reader_fuzzer.cc",
]
deps = [
+ "../../base:rtc_base_approved",
"../../modules/rtp_rtcp",
]
}
@@ -92,6 +96,7 @@
]
deps = [
"../../modules/rtp_rtcp",
+ "../../system_wrappers",
]
libfuzzer_options = [ "max_len=200" ]
}
@@ -101,6 +106,7 @@
"ulpfec_header_reader_fuzzer.cc",
]
deps = [
+ "../../base:rtc_base_approved",
"../../modules/rtp_rtcp",
"../../modules/rtp_rtcp:fec_test_helper",
]
@@ -122,6 +128,7 @@
"flexfec_receiver_fuzzer.cc",
]
deps = [
+ "../../base:rtc_base_approved",
"../../modules/rtp_rtcp",
]
libfuzzer_options = [ "max_len=2000" ]
@@ -133,6 +140,7 @@
]
deps = [
"../../modules/video_coding/",
+ "../../system_wrappers",
]
libfuzzer_options = [ "max_len=2000" ]
}
@@ -142,7 +150,9 @@
"rtcp_receiver_fuzzer.cc",
]
deps = [
+ "../../base:rtc_base_approved",
"../../modules/rtp_rtcp",
+ "../../system_wrappers:system_wrappers",
]
seed_corpus = "corpora/rtcp-corpus"
}
@@ -171,8 +181,11 @@
"congestion_controller_feedback_fuzzer.cc",
]
deps = [
+ "../../logging:rtc_event_log_api",
"../../logging:rtc_event_log_impl",
"../../modules/congestion_controller/",
+ "../../modules/remote_bitrate_estimator:remote_bitrate_estimator",
+ "../../modules/rtp_rtcp",
]
}
@@ -181,6 +194,12 @@
"audio_decoder_fuzzer.cc",
"audio_decoder_fuzzer.h",
]
+ deps = [
+ "../..:webrtc_common",
+ "../../api/audio_codecs:audio_codecs_api",
+ "../../base:rtc_base_approved",
+ "../../modules/rtp_rtcp",
+ ]
}
webrtc_fuzzer_test("audio_decoder_ilbc_fuzzer") {
@@ -260,6 +279,7 @@
"../../base:rtc_base_approved",
"../../base:rtc_base_tests_utils",
"../../modules/audio_coding:neteq",
+ "../../modules/audio_coding:neteq_test_tools",
"../../modules/audio_coding:neteq_tools_minimal",
"../../modules/audio_coding:pcm16b",
"../../modules/rtp_rtcp",
@@ -313,6 +333,7 @@
"pseudotcp_parser_fuzzer.cc",
]
deps = [
+ "../../base:rtc_base",
"../../p2p:rtc_p2p",
]
}
@@ -322,6 +343,7 @@
"transport_feedback_packet_loss_tracker_fuzzer.cc",
]
deps = [
+ "../../base:rtc_base_approved",
"../../modules/rtp_rtcp",
"../../voice_engine",
]