From bb08c3e29656fafe8a2d5d16ec4a62db49689f8a Mon Sep 17 00:00:00 2001 From: mbonadei Date: Wed, 26 Apr 2017 02:00:16 -0700 Subject: [PATCH] Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of https://codereview.webrtc.org/2838873002/ ) Reason for revert: Causes build problem: https://build.chromium.org/p/client.webrtc/builders/iOS64%20Sim%20Debug%20%28iOS%209.0%29/builds/1630/steps/generate%20build%20files%20%28mb%29/logs/stdio Original issue's description: > Creating webrtc/modules:module_api > > This target keeps track of .h the files under webrtc/modules/include/ > that are not part of any target. > If a .h file is not part of a target the 'gn check' utility is not > able to spot if a target is missing a dependency because even if > it parses '#include' directives it is not able to find a target that > contains these headers. > > BUG=webrtc:7513 > NOTRY=True > > Review-Url: https://codereview.webrtc.org/2838873002 > Cr-Commit-Position: refs/heads/master@{#17880} > Committed: https://chromium.googlesource.com/external/webrtc/+/5a1a092ed09ca92719eeb293275f64c0cdcc0e51 TBR=kjellander@webrtc.org # Skipping CQ checks because original CL landed less than 1 days ago. NOPRESUBMIT=true NOTREECHECKS=true NOTRY=true BUG=webrtc:7513 Review-Url: https://codereview.webrtc.org/2839963005 Cr-Commit-Position: refs/heads/master@{#17881} --- webrtc/api/BUILD.gn | 1 - webrtc/audio/utility/BUILD.gn | 2 -- webrtc/common_video/BUILD.gn | 1 - webrtc/modules/BUILD.gn | 15 ++------------- webrtc/modules/audio_coding/BUILD.gn | 15 +-------------- webrtc/modules/audio_conference_mixer/BUILD.gn | 1 - webrtc/modules/audio_device/BUILD.gn | 1 - webrtc/modules/audio_mixer/BUILD.gn | 3 --- webrtc/modules/audio_processing/BUILD.gn | 4 ---- webrtc/modules/congestion_controller/BUILD.gn | 1 - webrtc/modules/media_file/BUILD.gn | 1 - webrtc/modules/pacing/BUILD.gn | 1 - webrtc/modules/remote_bitrate_estimator/BUILD.gn | 1 - webrtc/modules/rtp_rtcp/BUILD.gn | 4 ---- webrtc/modules/utility/BUILD.gn | 2 -- webrtc/modules/video_capture/BUILD.gn | 1 - webrtc/modules/video_coding/BUILD.gn | 5 ----- webrtc/modules/video_processing/BUILD.gn | 4 ---- webrtc/sdk/BUILD.gn | 1 - webrtc/tools/BUILD.gn | 2 -- webrtc/video/BUILD.gn | 2 -- webrtc/voice_engine/BUILD.gn | 7 ------- 22 files changed, 3 insertions(+), 72 deletions(-) diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn index 3fa61a51b1..b8ad90505f 100644 --- a/webrtc/api/BUILD.gn +++ b/webrtc/api/BUILD.gn @@ -156,7 +156,6 @@ rtc_source_set("audio_mixer_api") { deps = [ "../base:rtc_base_approved", - "../modules:module_api", ] } diff --git a/webrtc/audio/utility/BUILD.gn b/webrtc/audio/utility/BUILD.gn index ac477e4f25..2ef5eba338 100644 --- a/webrtc/audio/utility/BUILD.gn +++ b/webrtc/audio/utility/BUILD.gn @@ -22,7 +22,6 @@ rtc_static_library("audio_frame_operations") { deps = [ "../..:webrtc_common", "../../base:rtc_base_approved", - "../../modules:module_api", "../../modules/audio_coding:audio_format_conversion", ] } @@ -36,7 +35,6 @@ if (rtc_include_tests) { deps = [ ":audio_frame_operations", "../../base:rtc_base_approved", - "../../modules:module_api", "../../test:test_support", "//testing/gtest", ] diff --git a/webrtc/common_video/BUILD.gn b/webrtc/common_video/BUILD.gn index 6b7eb01400..152f98b089 100644 --- a/webrtc/common_video/BUILD.gn +++ b/webrtc/common_video/BUILD.gn @@ -59,7 +59,6 @@ rtc_static_library("common_video") { "..:webrtc_common", "../base:rtc_base", "../base:rtc_task_queue", - "../modules:module_api", "../system_wrappers", ] public_deps = [ diff --git a/webrtc/modules/BUILD.gn b/webrtc/modules/BUILD.gn index 88752ebf16..e750a81097 100644 --- a/webrtc/modules/BUILD.gn +++ b/webrtc/modules/BUILD.gn @@ -29,18 +29,6 @@ group("modules") { ] } -rtc_source_set("module_api") { - sources = [ - "include/module.h", - "include/module_common_types.h", - ] - deps = [ - "..:webrtc_common", - "../api:video_frame_api", - "../base:rtc_base_approved", - ] -} - if (rtc_include_tests) { modules_tests_resources = [ "//resources/audio_coding/testfile32kHz.pcm", @@ -211,6 +199,8 @@ if (rtc_include_tests) { rtc_test("modules_unittests") { testonly = true + + deps = [] defines = [] sources = [ "module_common_types_unittest.cc", @@ -222,7 +212,6 @@ if (rtc_include_tests) { } deps += [ - ":module_api", "../test:test_main", "audio_coding:audio_coding_unittests", "audio_conference_mixer:audio_conference_mixer_unittests", diff --git a/webrtc/modules/audio_coding/BUILD.gn b/webrtc/modules/audio_coding/BUILD.gn index 8195e47aa6..4a2fdcaf87 100644 --- a/webrtc/modules/audio_coding/BUILD.gn +++ b/webrtc/modules/audio_coding/BUILD.gn @@ -127,7 +127,6 @@ rtc_source_set("audio_coding_module_typedefs") { "include/audio_coding_module_typedefs.h", ] deps = [ - "..:module_api", "../..:webrtc_common", ] } @@ -164,7 +163,6 @@ rtc_static_library("audio_coding") { } deps = audio_coding_deps + [ - "..:module_api", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", ":audio_coding_module_typedefs", @@ -1069,7 +1067,6 @@ rtc_static_library("neteq") { ":isac_fix", ":neteq_decoder_enum", ":pcm16b", - "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../base:gtest_prod", @@ -1201,7 +1198,6 @@ if (rtc_include_tests) { ":audio_coding_module_typedefs", ":audio_format_conversion", ":pcm16b_c", - "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../base:rtc_base_approved", @@ -1305,7 +1301,6 @@ if (rtc_include_tests) { ":audio_coding", ":audio_coding_module_typedefs", ":audio_format_conversion", - "..:module_api", "../../:webrtc_common", "../../base:rtc_base_approved", "../../system_wrappers", @@ -1333,7 +1328,6 @@ if (rtc_include_tests) { deps = [ ":audio_coding", ":audio_format_conversion", - "..:module_api", "../../:webrtc_common", "../../base:rtc_base_approved", "../../system_wrappers", @@ -1435,9 +1429,7 @@ if (rtc_include_tests) { rtc_test("neteq_rtpplay") { testonly = true defines = [] - deps = [ - "..:module_api", - ] + deps = [] sources = [ "neteq/tools/neteq_rtpplay.cc", ] @@ -1517,7 +1509,6 @@ if (rtc_include_tests) { ":neteq", ":neteq_unittest_tools", ":pcm16b", - "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", @@ -1543,7 +1534,6 @@ if (rtc_include_tests) { deps = [ ":neteq", ":neteq_unittest_tools", - "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:builtin_audio_decoder_factory", "../../base:rtc_base_approved", @@ -1598,7 +1588,6 @@ if (rtc_include_tests) { deps = [ ":audio_encoder_interface", ":pcm16b", - "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../base:rtc_base_approved", @@ -1636,7 +1625,6 @@ if (rtc_include_tests) { ":ilbc", ":isac", ":pcm16b", - "..:module_api", "../..:webrtc_common", "//testing/gtest", ] @@ -2148,7 +2136,6 @@ if (rtc_include_tests) { ":red", ":rent_a_codec", ":webrtc_opus", - "..:module_api", "../..:webrtc_common", "../../api/audio_codecs:audio_codecs_api", "../../api/audio_codecs:builtin_audio_decoder_factory", diff --git a/webrtc/modules/audio_conference_mixer/BUILD.gn b/webrtc/modules/audio_conference_mixer/BUILD.gn index 8939da222e..fc9904c23b 100644 --- a/webrtc/modules/audio_conference_mixer/BUILD.gn +++ b/webrtc/modules/audio_conference_mixer/BUILD.gn @@ -39,7 +39,6 @@ rtc_static_library("audio_conference_mixer") { } deps = [ - "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", "../../base:rtc_base_approved", diff --git a/webrtc/modules/audio_device/BUILD.gn b/webrtc/modules/audio_device/BUILD.gn index ab0b4f5b32..1e691fa9bc 100644 --- a/webrtc/modules/audio_device/BUILD.gn +++ b/webrtc/modules/audio_device/BUILD.gn @@ -49,7 +49,6 @@ rtc_static_library("audio_device") { public_configs = [ ":audio_device_config" ] deps = [ - "..:module_api", "../..:webrtc_common", "../../base:rtc_base_approved", "../../base:rtc_task_queue", diff --git a/webrtc/modules/audio_mixer/BUILD.gn b/webrtc/modules/audio_mixer/BUILD.gn index cd3b768ee1..d8acc05a4a 100644 --- a/webrtc/modules/audio_mixer/BUILD.gn +++ b/webrtc/modules/audio_mixer/BUILD.gn @@ -38,7 +38,6 @@ rtc_static_library("audio_mixer_impl") { deps = [ ":audio_frame_manipulator", - "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", "../../base:rtc_base_approved", @@ -59,7 +58,6 @@ rtc_static_library("audio_frame_manipulator") { ] deps = [ - "..:module_api", "../../audio/utility", "../../base:rtc_base_approved", ] @@ -87,7 +85,6 @@ if (rtc_include_tests) { deps = [ ":audio_frame_manipulator", ":audio_mixer_impl", - "..:module_api", "../../api:audio_mixer_api", "../../audio/utility:audio_frame_operations", "../../base:rtc_base", diff --git a/webrtc/modules/audio_processing/BUILD.gn b/webrtc/modules/audio_processing/BUILD.gn index ff9a4d6474..0f6de09052 100644 --- a/webrtc/modules/audio_processing/BUILD.gn +++ b/webrtc/modules/audio_processing/BUILD.gn @@ -230,7 +230,6 @@ rtc_static_library("audio_processing") { defines = [] deps = [ - "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", "../../base:gtest_prod", @@ -532,7 +531,6 @@ if (rtc_include_tests) { deps = [ ":audio_processing", ":audioproc_test_utils", - "..:module_api", "../..:webrtc_common", "../../base:gtest_prod", "../../base:protobuf_utils", @@ -751,7 +749,6 @@ if (rtc_include_tests) { deps = [ ":audio_processing", - "..:module_api", "../../base:rtc_base_approved", "../../common_audio", "../../system_wrappers:system_wrappers", @@ -767,7 +764,6 @@ if (rtc_include_tests) { ] deps = [ ":audio_processing", - "..:module_api", "../..:webrtc_common", "../../common_audio:common_audio", "../../system_wrappers:metrics_default", diff --git a/webrtc/modules/congestion_controller/BUILD.gn b/webrtc/modules/congestion_controller/BUILD.gn index 3dcbb27b24..647079a382 100644 --- a/webrtc/modules/congestion_controller/BUILD.gn +++ b/webrtc/modules/congestion_controller/BUILD.gn @@ -45,7 +45,6 @@ rtc_static_library("congestion_controller") { } deps = [ - "..:module_api", "../..:webrtc_common", "../../base:rtc_base", "../../base:rtc_base_approved", diff --git a/webrtc/modules/media_file/BUILD.gn b/webrtc/modules/media_file/BUILD.gn index 7ab897f28f..4f8fbbc4e2 100644 --- a/webrtc/modules/media_file/BUILD.gn +++ b/webrtc/modules/media_file/BUILD.gn @@ -33,7 +33,6 @@ rtc_static_library("media_file") { } deps = [ - "..:module_api", "../..:webrtc_common", "../../base:rtc_base_approved", "../../common_audio", diff --git a/webrtc/modules/pacing/BUILD.gn b/webrtc/modules/pacing/BUILD.gn index 57126d7143..ce2356e28a 100644 --- a/webrtc/modules/pacing/BUILD.gn +++ b/webrtc/modules/pacing/BUILD.gn @@ -26,7 +26,6 @@ rtc_static_library("pacing") { } deps = [ - "..:module_api", "../../:webrtc_common", "../../base:rtc_base_approved", "../../logging:rtc_event_log_api", diff --git a/webrtc/modules/remote_bitrate_estimator/BUILD.gn b/webrtc/modules/remote_bitrate_estimator/BUILD.gn index 04f2f7cc55..c2e5d31492 100644 --- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn +++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn @@ -109,7 +109,6 @@ if (rtc_include_tests) { deps = [ ":remote_bitrate_estimator", - "..:module_api", "../..:webrtc_common", "../../base:gtest_prod", "../../base:rtc_base", diff --git a/webrtc/modules/rtp_rtcp/BUILD.gn b/webrtc/modules/rtp_rtcp/BUILD.gn index 5d754d9974..a369218a68 100644 --- a/webrtc/modules/rtp_rtcp/BUILD.gn +++ b/webrtc/modules/rtp_rtcp/BUILD.gn @@ -166,7 +166,6 @@ rtc_static_library("rtp_rtcp") { } deps = [ - "..:module_api", "../..:webrtc_common", "../../api:libjingle_peerconnection_api", "../../api:transport_api", @@ -201,7 +200,6 @@ rtc_source_set("fec_test_helper") { ] deps = [ ":rtp_rtcp", - "..:module_api", "../../base:rtc_base_approved", ] @@ -259,7 +257,6 @@ if (rtc_include_tests) { ] deps = [ ":rtp_rtcp", - "..:module_api", "../../base:rtc_base_approved", "../../test:test_support", ] @@ -339,7 +336,6 @@ if (rtc_include_tests) { ":fec_test_helper", ":mock_rtp_rtcp", ":rtp_rtcp", - "..:module_api", "../..:webrtc_common", "../../api:transport_api", "../../base:rtc_base_approved", diff --git a/webrtc/modules/utility/BUILD.gn b/webrtc/modules/utility/BUILD.gn index 71238905dc..3d32ac27d5 100644 --- a/webrtc/modules/utility/BUILD.gn +++ b/webrtc/modules/utility/BUILD.gn @@ -30,7 +30,6 @@ rtc_static_library("utility") { } deps = [ - "..:module_api", "../..:webrtc_common", "../../audio/utility:audio_frame_operations", "../../base:rtc_task_queue", @@ -55,7 +54,6 @@ if (rtc_include_tests) { ] deps = [ ":utility", - "..:module_api", "../../base:rtc_task_queue", "../../test:test_support", "//testing/gmock", diff --git a/webrtc/modules/video_capture/BUILD.gn b/webrtc/modules/video_capture/BUILD.gn index c902ee86ff..b7482a2555 100644 --- a/webrtc/modules/video_capture/BUILD.gn +++ b/webrtc/modules/video_capture/BUILD.gn @@ -26,7 +26,6 @@ rtc_static_library("video_capture_module") { ] deps = [ - "..:module_api", "../..:webrtc_common", "../../base:rtc_base_approved", "../../common_video", diff --git a/webrtc/modules/video_coding/BUILD.gn b/webrtc/modules/video_coding/BUILD.gn index 477e06444e..bafc8bae07 100644 --- a/webrtc/modules/video_coding/BUILD.gn +++ b/webrtc/modules/video_coding/BUILD.gn @@ -94,7 +94,6 @@ rtc_static_library("video_coding") { ":webrtc_i420", ":webrtc_vp8", ":webrtc_vp9", - "..:module_api", "../..:video_stream_api", "../..:webrtc_common", "../../base:rtc_base", @@ -130,7 +129,6 @@ rtc_static_library("video_coding_utility") { } deps = [ - "..:module_api", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", "../../base:rtc_base_approved", @@ -227,7 +225,6 @@ rtc_static_library("webrtc_vp8") { deps = [ ":video_coding_utility", - "..:module_api", "../..:webrtc_common", "../../api/video_codecs:video_codecs_api", "../../base:rtc_base_approved", @@ -263,7 +260,6 @@ rtc_static_library("webrtc_vp9") { deps = [ ":video_coding_utility", - "..:module_api", "../../base:rtc_base_approved", "../../common_video", "../../system_wrappers", @@ -547,7 +543,6 @@ if (rtc_include_tests) { ":webrtc_h264", ":webrtc_vp8", ":webrtc_vp9", - "..:module_api", "../..:webrtc_common", "../../api:video_frame_api", "../../api/video_codecs:video_codecs_api", diff --git a/webrtc/modules/video_processing/BUILD.gn b/webrtc/modules/video_processing/BUILD.gn index c4c9c3b894..7c9391a6d4 100644 --- a/webrtc/modules/video_processing/BUILD.gn +++ b/webrtc/modules/video_processing/BUILD.gn @@ -26,7 +26,6 @@ rtc_static_library("video_processing") { deps = [ ":denoiser_filter", - "..:module_api", "../../base:rtc_base_approved", "../../common_audio", "../../common_video", @@ -52,9 +51,6 @@ rtc_source_set("denoiser_filter") { sources = [ "util/denoiser_filter.h", ] - deps = [ - "..:module_api", - ] } if (build_video_processing_sse2) { diff --git a/webrtc/sdk/BUILD.gn b/webrtc/sdk/BUILD.gn index ca6054e0c9..a352f9c65a 100644 --- a/webrtc/sdk/BUILD.gn +++ b/webrtc/sdk/BUILD.gn @@ -423,7 +423,6 @@ if (is_ios || is_mac) { "../base:rtc_base_approved", "../common_video", "../media:rtc_media_base", - "../modules:module_api", "../modules/video_coding:video_coding_utility", "../modules/video_coding:webrtc_h264", "../system_wrappers", diff --git a/webrtc/tools/BUILD.gn b/webrtc/tools/BUILD.gn index 2850552f1a..be3296d1ff 100644 --- a/webrtc/tools/BUILD.gn +++ b/webrtc/tools/BUILD.gn @@ -210,7 +210,6 @@ if (rtc_enable_protobuf) { "../call:call_interfaces", "../logging:rtc_event_log_impl", "../logging:rtc_event_log_parser", - "../modules:module_api", "../modules/audio_coding:ana_debug_dump_proto", # TODO(kwiberg): Remove this dependency. @@ -262,7 +261,6 @@ if (rtc_include_tests) { } deps = [ - "../modules:module_api", "../modules/audio_processing", "../system_wrappers:metrics_default", "../test:test_support", diff --git a/webrtc/video/BUILD.gn b/webrtc/video/BUILD.gn index 61d628ad74..76f6ece715 100644 --- a/webrtc/video/BUILD.gn +++ b/webrtc/video/BUILD.gn @@ -65,7 +65,6 @@ rtc_static_library("video") { "../common_video", "../logging:rtc_event_log_api", "../media:rtc_media_base", - "../modules:module_api", "../modules/bitrate_controller", "../modules/congestion_controller", "../modules/pacing", @@ -261,7 +260,6 @@ if (rtc_include_tests) { "../logging:rtc_event_log_api", "../media:rtc_media_base", "../media:rtc_media_tests_utils", - "../modules:module_api", "../modules/pacing", "../modules/rtp_rtcp", "../modules/rtp_rtcp:mock_rtp_rtcp", diff --git a/webrtc/voice_engine/BUILD.gn b/webrtc/voice_engine/BUILD.gn index be6faabe39..ca774f2800 100644 --- a/webrtc/voice_engine/BUILD.gn +++ b/webrtc/voice_engine/BUILD.gn @@ -16,7 +16,6 @@ rtc_static_library("audio_coder") { deps = [ "..:webrtc_common", "../api/audio_codecs:builtin_audio_decoder_factory", - "../modules:module_api", "../modules/audio_coding", "../modules/audio_coding:audio_encoder_factory_interface", "../modules/audio_coding:audio_format_conversion", @@ -40,7 +39,6 @@ rtc_static_library("file_player") { "..:webrtc_common", "../base:rtc_base_approved", "../common_audio", - "../modules:module_api", "../modules/media_file", ] @@ -60,7 +58,6 @@ rtc_static_library("file_recorder") { "..:webrtc_common", "../base:rtc_base_approved", "../common_audio", - "../modules:module_api", "../modules/media_file:media_file", "../system_wrappers", ] @@ -144,7 +141,6 @@ rtc_static_library("voice_engine") { "../audio/utility:audio_frame_operations", "../base:rtc_base_approved", "../base:rtc_task_queue", - "../modules:module_api", # TODO(nisse): Delete when declaration of RtpTransportController # and related interfaces move to api/. @@ -176,7 +172,6 @@ rtc_static_library("audio_level") { "..:webrtc_common", "../base:rtc_base_approved", "../common_audio", - "../modules:module_api", ] } @@ -186,7 +181,6 @@ if (rtc_include_tests) { ":file_player", ":voice_engine", "../base:rtc_base_approved", - "../modules:module_api", "../test:test_common", "//testing/gmock", "//testing/gtest", @@ -250,7 +244,6 @@ if (rtc_include_tests) { ":voice_engine", "..:webrtc_common", "../base:rtc_base_approved", - "../modules:module_api", "../modules/audio_device:audio_device", "../modules/audio_processing:audio_processing", "../modules/rtp_rtcp:rtp_rtcp",