Revert of GN: Refactor modules_unittests to eliminate package boundary violations. (patchset #4 id:130001 of https://codereview.webrtc.org/2649563002/ )

Reason for revert:
Did break the bots.
https://build.chromium.org/p/client.webrtc/builders/iOS32%20Release/builds/9807

Original issue's description:
> GN: Refactor modules_unittests to eliminate package boundary violations.
>
> Also move bwe_simulator to webrtc/modules/remote_bitrate_estimator
>
> BUG=webrtc:6954
> NOTRY=True
>
> Review-Url: https://codereview.webrtc.org/2649563002
> Cr-Commit-Position: refs/heads/master@{#16270}
> Committed: 36cb55d715

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
BUG=webrtc:6954

Review-Url: https://codereview.webrtc.org/2651023005
Cr-Commit-Position: refs/heads/master@{#16271}
This commit is contained in:
ehmaldonado 2017-01-25 08:11:28 -08:00 committed by Commit bot
parent 36cb55d715
commit 3373eaa577
6 changed files with 83 additions and 147 deletions

View File

@ -51,16 +51,62 @@ if (rtc_include_tests) {
rtc_test("modules_tests") {
testonly = true
videoprocessor_defines = []
if (rtc_use_h264) {
videoprocessor_defines += [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ]
}
defines = audio_coding_defines + videoprocessor_defines
deps = [
"..:webrtc_common",
"../common_video",
"../media:rtc_media_base",
"../modules/audio_coding",
"../modules/audio_coding:audio_format_conversion",
"../modules/rtp_rtcp",
"../modules/utility",
"../modules/video_coding",
"../modules/video_coding:video_codecs_test_framework",
"../system_wrappers",
"../test:test_main",
"audio_coding:audio_coding_modules_tests",
"desktop_capture:desktop_capture_modules_tests",
"rtp_rtcp:rtp_rtcp_modules_tests",
"video_coding:video_coding_modules_tests",
"//testing/gmock",
"//testing/gtest",
]
sources = [
"audio_coding/test/APITest.cc",
"audio_coding/test/Channel.cc",
"audio_coding/test/EncodeDecodeTest.cc",
"audio_coding/test/PCMFile.cc",
"audio_coding/test/PacketLossTest.cc",
"audio_coding/test/RTPFile.cc",
"audio_coding/test/TestAllCodecs.cc",
"audio_coding/test/TestRedFec.cc",
"audio_coding/test/TestStereo.cc",
"audio_coding/test/TestVADDTX.cc",
"audio_coding/test/Tester.cc",
"audio_coding/test/TwoWayCommunication.cc",
"audio_coding/test/iSACTest.cc",
"audio_coding/test/opus_test.cc",
"audio_coding/test/target_delay_unittest.cc",
"audio_coding/test/utility.cc",
"rtp_rtcp/test/testFec/test_fec.cc",
"video_coding/codecs/test/videoprocessor_integrationtest.cc",
"video_coding/codecs/vp8/test/vp8_impl_unittest.cc",
]
if (rtc_desktop_capture_supported) {
deps += [
"desktop_capture:desktop_capture_mock",
"desktop_capture:screen_drawer",
]
sources += [
"desktop_capture/screen_capturer_integration_test.cc",
"desktop_capture/screen_drawer_unittest.cc",
]
}
data = modules_tests_resources
if (is_android) {
@ -246,4 +292,37 @@ if (rtc_include_tests) {
ldflags = [ "-ObjC" ]
}
}
rtc_test("bwe_simulator") {
testonly = true
deps = []
sources = [
"remote_bitrate_estimator/bwe_simulations.cc",
]
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" ]
}
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
deps += [
"..:webrtc_common",
"../base:rtc_base_approved",
"../test:test_common",
"../test:test_main",
"remote_bitrate_estimator:bwe_simulator_lib",
"video_capture",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
}
}

View File

@ -1139,50 +1139,6 @@ if (rtc_include_tests) {
}
}
rtc_source_set("audio_coding_modules_tests") {
testonly = true
sources = [
"test/APITest.cc",
"test/Channel.cc",
"test/EncodeDecodeTest.cc",
"test/PCMFile.cc",
"test/PacketLossTest.cc",
"test/RTPFile.cc",
"test/TestAllCodecs.cc",
"test/TestRedFec.cc",
"test/TestStereo.cc",
"test/TestVADDTX.cc",
"test/Tester.cc",
"test/TwoWayCommunication.cc",
"test/iSACTest.cc",
"test/opus_test.cc",
"test/target_delay_unittest.cc",
"test/utility.cc",
]
deps = [
":audio_coding",
":audio_format_conversion",
":builtin_audio_decoder_factory",
":pcm16b_c",
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:fileutils",
"../../test:test_support",
]
defines = audio_coding_defines
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
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" ]
}
}
rtc_source_set("audio_coding_perf_tests") {
testonly = true
sources = [

View File

@ -33,27 +33,6 @@ rtc_static_library("primitives") {
}
if (rtc_include_tests) {
rtc_source_set("desktop_capture_modules_tests") {
testonly = true
sources = []
deps = []
if (rtc_desktop_capture_supported) {
deps += [
":desktop_capture_mock",
":primitives",
":screen_drawer",
"../../base:rtc_base",
"../../base:rtc_base_approved",
"../../system_wrappers:system_wrappers",
"../../test:test_support",
]
sources += [
"screen_capturer_integration_test.cc",
"screen_drawer_unittest.cc",
]
}
}
rtc_source_set("desktop_capture_unittests") {
testonly = true
sources = []

View File

@ -185,36 +185,4 @@ if (rtc_include_tests) {
suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
}
}
rtc_test("bwe_simulator") {
testonly = true
sources = [
"bwe_simulations.cc",
]
deps = [
":bwe_simulator_lib",
":remote_bitrate_estimator",
"../..:webrtc_common",
"../../base:rtc_base_approved",
"../../test:test_common",
"../../test:test_main",
"../video_capture",
"//testing/gmock",
"//testing/gtest",
"//third_party/gflags",
]
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" ]
}
if (is_win) {
cflags = [
# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
"/wd4373", # virtual function override.
]
}
}
}

View File

@ -211,22 +211,6 @@ if (rtc_include_tests) {
]
} # test_packet_masks_metrics
rtc_source_set("rtp_rtcp_modules_tests") {
testonly = true
sources = [
"test/testFec/test_fec.cc",
]
deps = [
":rtp_rtcp",
"../../base:rtc_base_approved",
"../../test:test_support",
]
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" ]
}
}
rtc_source_set("rtp_rtcp_unittests") {
testonly = true
sources = [

View File

@ -310,36 +310,6 @@ if (rtc_include_tests) {
"../../test:test_support",
]
}
rtc_source_set("video_coding_modules_tests") {
testonly = true
sources = [
"codecs/test/videoprocessor_integrationtest.cc",
"codecs/vp8/test/vp8_impl_unittest.cc",
]
deps = [
":video_codecs_test_framework",
":video_coding",
":webrtc_h264",
":webrtc_vp8",
":webrtc_vp9",
"../..:webrtc_common",
"../../api:video_frame_api",
"../../base:rtc_base_approved",
"../../common_video:common_video",
"../../media:rtc_media_base",
"../../test:test_support",
"../../test:video_test_common",
]
if (rtc_use_h264) {
defines = [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ]
}
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" ]
}
}
rtc_source_set("video_coding_unittests") {
testonly = true
sources = [