Remove webrtc_tests.
This target is just confusing, since it runs essentially random tests out of the test binaries it's depending on. This CL makes the top GN file structured like Chromium's is. A possible secondary function is to act as a link test (maybe to catch twice-defined symbols etc), but that seems highly speculative. Bug: None Change-Id: Ic8a36ffe96408312e602a58f8806b0694fedaad6 Reviewed-on: https://webrtc-review.googlesource.com/27380 Commit-Queue: Patrik Höglund <phoglund@webrtc.org> Reviewed-by: Oleh Prypin <oprypin@webrtc.org> Cr-Commit-Position: refs/heads/master@{#21028}
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101
BUILD.gn
101
BUILD.gn
@ -6,6 +6,12 @@
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# in the file PATENTS. All contributing project authors may
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# be found in the AUTHORS file in the root of the source tree.
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# This is the root build file for GN. GN will start processing by loading this
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# file, and recursively load all dependencies until all dependencies are either
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# resolved or known not to exist (which will cause the build to fail). So if
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# you add a new build file, there must be some path of dependencies from this
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# file to your new one or GN won't know about it.
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import("//build/config/linux/pkg_config.gni")
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import("//build/config/sanitizers/sanitizers.gni")
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import("webrtc.gni")
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@ -16,6 +22,8 @@ if (is_android) {
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}
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if (!build_with_chromium) {
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# This target should (transitively) cause everything to be built; if you run
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# 'ninja default' and then 'ninja all', the second build should do no work.
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group("default") {
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testonly = true
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deps = [
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@ -24,7 +32,46 @@ if (!build_with_chromium) {
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"rtc_tools",
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]
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if (rtc_include_tests) {
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deps += [ ":webrtc_tests" ]
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deps += [
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":rtc_unittests",
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":video_engine_tests",
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":webrtc_nonparallel_tests",
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":webrtc_perf_tests",
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"common_audio:common_audio_unittests",
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"common_video:common_video_unittests",
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"media:rtc_media_unittests",
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"modules:modules_tests",
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"modules:modules_unittests",
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"modules/audio_coding:audio_coding_tests",
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"modules/audio_processing:audio_processing_tests",
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"modules/remote_bitrate_estimator:bwe_simulations_tests",
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"modules/rtp_rtcp:test_packet_masks_metrics",
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"modules/video_capture:video_capture_internal_impl",
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"ortc:ortc_unittests",
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"pc:peerconnection_unittests",
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"pc:rtc_pc_unittests",
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"rtc_base:rtc_base_tests_utils",
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"stats:rtc_stats_unittests",
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"system_wrappers:system_wrappers_unittests",
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"test",
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"video:screenshare_loopback",
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"video:video_loopback",
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"voice_engine:voice_engine_unittests",
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]
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if (is_android) {
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deps += [
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":android_junit_tests",
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"sdk/android:libjingle_peerconnection_android_unittest",
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]
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} else {
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deps += [ "modules/video_capture:video_capture_tests" ]
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}
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if (rtc_enable_protobuf) {
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deps += [
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"audio:low_bandwidth_audio_test",
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"logging:rtc_event_log2rtp_dump",
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]
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}
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}
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}
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}
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@ -294,58 +341,6 @@ if (!build_with_chromium) {
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deps += [ "logging:rtc_event_log_proto" ]
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}
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}
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if (rtc_include_tests) {
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# Target to build all the WebRTC tests (but not examples or tools).
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# Executable in order to get a target that links all WebRTC code.
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rtc_executable("webrtc_tests") {
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testonly = true
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# Only the root target should depend on this.
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visibility = [ "//:default" ]
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deps = [
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":rtc_unittests",
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":video_engine_tests",
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":webrtc_nonparallel_tests",
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":webrtc_perf_tests",
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"common_audio:common_audio_unittests",
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"common_video:common_video_unittests",
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"media:rtc_media_unittests",
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"modules:modules_tests",
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"modules:modules_unittests",
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"modules/audio_coding:audio_coding_tests",
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"modules/audio_processing:audio_processing_tests",
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"modules/remote_bitrate_estimator:bwe_simulations_tests",
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"modules/rtp_rtcp:test_packet_masks_metrics",
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"modules/video_capture:video_capture_internal_impl",
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"ortc:ortc_unittests",
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"pc:peerconnection_unittests",
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"pc:rtc_pc_unittests",
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"rtc_base:rtc_base_tests_utils",
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"stats:rtc_stats_unittests",
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"system_wrappers:system_wrappers_unittests",
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"test",
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"video:screenshare_loopback",
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"video:video_loopback",
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"voice_engine:voice_engine_unittests",
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]
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if (is_android) {
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deps += [
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":android_junit_tests",
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"sdk/android:libjingle_peerconnection_android_unittest",
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]
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} else {
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deps += [ "modules/video_capture:video_capture_tests" ]
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}
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if (rtc_enable_protobuf) {
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deps += [
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"audio:low_bandwidth_audio_test",
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"logging:rtc_event_log2rtp_dump",
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]
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}
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}
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}
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}
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rtc_static_library("webrtc_common") {
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