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:
parent
36cb55d715
commit
3373eaa577
@ -51,16 +51,62 @@ if (rtc_include_tests) {
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rtc_test("modules_tests") {
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testonly = true
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videoprocessor_defines = []
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if (rtc_use_h264) {
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videoprocessor_defines += [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ]
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}
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defines = audio_coding_defines + videoprocessor_defines
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deps = [
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"..:webrtc_common",
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"../common_video",
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"../media:rtc_media_base",
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"../modules/audio_coding",
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"../modules/audio_coding:audio_format_conversion",
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"../modules/rtp_rtcp",
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"../modules/utility",
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"../modules/video_coding",
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"../modules/video_coding:video_codecs_test_framework",
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"../system_wrappers",
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"../test:test_main",
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"audio_coding:audio_coding_modules_tests",
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"desktop_capture:desktop_capture_modules_tests",
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"rtp_rtcp:rtp_rtcp_modules_tests",
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"video_coding:video_coding_modules_tests",
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"//testing/gmock",
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"//testing/gtest",
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]
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sources = [
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"audio_coding/test/APITest.cc",
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"audio_coding/test/Channel.cc",
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"audio_coding/test/EncodeDecodeTest.cc",
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"audio_coding/test/PCMFile.cc",
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"audio_coding/test/PacketLossTest.cc",
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"audio_coding/test/RTPFile.cc",
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"audio_coding/test/TestAllCodecs.cc",
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"audio_coding/test/TestRedFec.cc",
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"audio_coding/test/TestStereo.cc",
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"audio_coding/test/TestVADDTX.cc",
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"audio_coding/test/Tester.cc",
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"audio_coding/test/TwoWayCommunication.cc",
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"audio_coding/test/iSACTest.cc",
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"audio_coding/test/opus_test.cc",
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"audio_coding/test/target_delay_unittest.cc",
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"audio_coding/test/utility.cc",
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"rtp_rtcp/test/testFec/test_fec.cc",
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"video_coding/codecs/test/videoprocessor_integrationtest.cc",
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"video_coding/codecs/vp8/test/vp8_impl_unittest.cc",
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]
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if (rtc_desktop_capture_supported) {
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deps += [
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"desktop_capture:desktop_capture_mock",
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"desktop_capture:screen_drawer",
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]
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sources += [
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"desktop_capture/screen_capturer_integration_test.cc",
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"desktop_capture/screen_drawer_unittest.cc",
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]
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}
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data = modules_tests_resources
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if (is_android) {
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@ -246,4 +292,37 @@ if (rtc_include_tests) {
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ldflags = [ "-ObjC" ]
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}
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}
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rtc_test("bwe_simulator") {
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testonly = true
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deps = []
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sources = [
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"remote_bitrate_estimator/bwe_simulations.cc",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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if (is_win) {
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cflags = [
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# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
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"/wd4373", # virtual function override.
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]
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}
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deps += [
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"..:webrtc_common",
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"../base:rtc_base_approved",
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"../test:test_common",
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"../test:test_main",
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"remote_bitrate_estimator:bwe_simulator_lib",
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"video_capture",
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"//testing/gmock",
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"//testing/gtest",
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"//third_party/gflags",
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]
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}
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}
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@ -1139,50 +1139,6 @@ if (rtc_include_tests) {
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}
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}
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rtc_source_set("audio_coding_modules_tests") {
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testonly = true
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sources = [
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"test/APITest.cc",
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"test/Channel.cc",
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"test/EncodeDecodeTest.cc",
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"test/PCMFile.cc",
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"test/PacketLossTest.cc",
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"test/RTPFile.cc",
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"test/TestAllCodecs.cc",
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"test/TestRedFec.cc",
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"test/TestStereo.cc",
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"test/TestVADDTX.cc",
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"test/Tester.cc",
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"test/TwoWayCommunication.cc",
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"test/iSACTest.cc",
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"test/opus_test.cc",
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"test/target_delay_unittest.cc",
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"test/utility.cc",
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]
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deps = [
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":audio_coding",
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":audio_format_conversion",
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":builtin_audio_decoder_factory",
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":pcm16b_c",
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"../..:webrtc_common",
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"../../base:rtc_base_approved",
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"../../system_wrappers:system_wrappers",
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"../../test:fileutils",
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"../../test:test_support",
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]
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defines = audio_coding_defines
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if (is_win) {
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cflags = [
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# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
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"/wd4373", # virtual function override.
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]
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}
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("audio_coding_perf_tests") {
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testonly = true
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sources = [
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@ -33,27 +33,6 @@ rtc_static_library("primitives") {
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}
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if (rtc_include_tests) {
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rtc_source_set("desktop_capture_modules_tests") {
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testonly = true
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sources = []
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deps = []
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if (rtc_desktop_capture_supported) {
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deps += [
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":desktop_capture_mock",
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":primitives",
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":screen_drawer",
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"../../base:rtc_base",
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"../../base:rtc_base_approved",
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"../../system_wrappers:system_wrappers",
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"../../test:test_support",
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]
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sources += [
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"screen_capturer_integration_test.cc",
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"screen_drawer_unittest.cc",
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]
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}
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}
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rtc_source_set("desktop_capture_unittests") {
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testonly = true
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sources = []
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@ -185,36 +185,4 @@ if (rtc_include_tests) {
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_test("bwe_simulator") {
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testonly = true
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sources = [
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"bwe_simulations.cc",
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]
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deps = [
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":bwe_simulator_lib",
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":remote_bitrate_estimator",
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"../..:webrtc_common",
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"../../base:rtc_base_approved",
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"../../test:test_common",
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"../../test:test_main",
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"../video_capture",
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"//testing/gmock",
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"//testing/gtest",
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"//third_party/gflags",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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if (is_win) {
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cflags = [
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# TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
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"/wd4373", # virtual function override.
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]
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}
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}
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}
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@ -211,22 +211,6 @@ if (rtc_include_tests) {
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]
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} # test_packet_masks_metrics
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rtc_source_set("rtp_rtcp_modules_tests") {
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testonly = true
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sources = [
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"test/testFec/test_fec.cc",
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]
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deps = [
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":rtp_rtcp",
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"../../base:rtc_base_approved",
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"../../test:test_support",
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]
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("rtp_rtcp_unittests") {
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testonly = true
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sources = [
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@ -310,36 +310,6 @@ if (rtc_include_tests) {
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"../../test:test_support",
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]
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}
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rtc_source_set("video_coding_modules_tests") {
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testonly = true
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sources = [
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"codecs/test/videoprocessor_integrationtest.cc",
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"codecs/vp8/test/vp8_impl_unittest.cc",
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]
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deps = [
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":video_codecs_test_framework",
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":video_coding",
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":webrtc_h264",
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":webrtc_vp8",
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":webrtc_vp9",
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"../..:webrtc_common",
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"../../api:video_frame_api",
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"../../base:rtc_base_approved",
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"../../common_video:common_video",
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"../../media:rtc_media_base",
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"../../test:test_support",
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"../../test:video_test_common",
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]
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if (rtc_use_h264) {
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defines = [ "WEBRTC_VIDEOPROCESSOR_H264_TESTS" ]
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}
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if (!build_with_chromium && is_clang) {
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# Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
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suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
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}
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}
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rtc_source_set("video_coding_unittests") {
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testonly = true
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sources = [
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