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}
diff --git a/webrtc/api/BUILD.gn b/webrtc/api/BUILD.gn
index 3fa61a5..b8ad905 100644
--- a/webrtc/api/BUILD.gn
+++ b/webrtc/api/BUILD.gn
@@ -156,7 +156,6 @@
deps = [
"../base:rtc_base_approved",
- "../modules:module_api",
]
}
diff --git a/webrtc/audio/utility/BUILD.gn b/webrtc/audio/utility/BUILD.gn
index ac477e4..2ef5eba 100644
--- a/webrtc/audio/utility/BUILD.gn
+++ b/webrtc/audio/utility/BUILD.gn
@@ -22,7 +22,6 @@
deps = [
"../..:webrtc_common",
"../../base:rtc_base_approved",
- "../../modules:module_api",
"../../modules/audio_coding:audio_format_conversion",
]
}
@@ -36,7 +35,6 @@
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 6b7eb01..152f98b 100644
--- a/webrtc/common_video/BUILD.gn
+++ b/webrtc/common_video/BUILD.gn
@@ -59,7 +59,6 @@
"..: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 88752eb..e750a81 100644
--- a/webrtc/modules/BUILD.gn
+++ b/webrtc/modules/BUILD.gn
@@ -29,18 +29,6 @@
]
}
-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 @@
rtc_test("modules_unittests") {
testonly = true
+
+ deps = []
defines = []
sources = [
"module_common_types_unittest.cc",
@@ -222,7 +212,6 @@
}
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 8195e47..4a2fdca 100644
--- a/webrtc/modules/audio_coding/BUILD.gn
+++ b/webrtc/modules/audio_coding/BUILD.gn
@@ -127,7 +127,6 @@
"include/audio_coding_module_typedefs.h",
]
deps = [
- "..:module_api",
"../..:webrtc_common",
]
}
@@ -164,7 +163,6 @@
}
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 @@
":isac_fix",
":neteq_decoder_enum",
":pcm16b",
- "..:module_api",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:gtest_prod",
@@ -1201,7 +1198,6 @@
":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 @@
":audio_coding",
":audio_coding_module_typedefs",
":audio_format_conversion",
- "..:module_api",
"../../:webrtc_common",
"../../base:rtc_base_approved",
"../../system_wrappers",
@@ -1333,7 +1328,6 @@
deps = [
":audio_coding",
":audio_format_conversion",
- "..:module_api",
"../../:webrtc_common",
"../../base:rtc_base_approved",
"../../system_wrappers",
@@ -1435,9 +1429,7 @@
rtc_test("neteq_rtpplay") {
testonly = true
defines = []
- deps = [
- "..:module_api",
- ]
+ deps = []
sources = [
"neteq/tools/neteq_rtpplay.cc",
]
@@ -1517,7 +1509,6 @@
":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 @@
deps = [
":neteq",
":neteq_unittest_tools",
- "..:module_api",
"../..:webrtc_common",
"../../api/audio_codecs:builtin_audio_decoder_factory",
"../../base:rtc_base_approved",
@@ -1598,7 +1588,6 @@
deps = [
":audio_encoder_interface",
":pcm16b",
- "..:module_api",
"../..:webrtc_common",
"../../api/audio_codecs:audio_codecs_api",
"../../base:rtc_base_approved",
@@ -1636,7 +1625,6 @@
":ilbc",
":isac",
":pcm16b",
- "..:module_api",
"../..:webrtc_common",
"//testing/gtest",
]
@@ -2148,7 +2136,6 @@
":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 8939da2..fc9904c 100644
--- a/webrtc/modules/audio_conference_mixer/BUILD.gn
+++ b/webrtc/modules/audio_conference_mixer/BUILD.gn
@@ -39,7 +39,6 @@
}
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 ab0b4f5..1e691fa 100644
--- a/webrtc/modules/audio_device/BUILD.gn
+++ b/webrtc/modules/audio_device/BUILD.gn
@@ -49,7 +49,6 @@
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 cd3b768..d8acc05 100644
--- a/webrtc/modules/audio_mixer/BUILD.gn
+++ b/webrtc/modules/audio_mixer/BUILD.gn
@@ -38,7 +38,6 @@
deps = [
":audio_frame_manipulator",
- "..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../../base:rtc_base_approved",
@@ -59,7 +58,6 @@
]
deps = [
- "..:module_api",
"../../audio/utility",
"../../base:rtc_base_approved",
]
@@ -87,7 +85,6 @@
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 ff9a4d6..0f6de09 100644
--- a/webrtc/modules/audio_processing/BUILD.gn
+++ b/webrtc/modules/audio_processing/BUILD.gn
@@ -230,7 +230,6 @@
defines = []
deps = [
- "..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../../base:gtest_prod",
@@ -532,7 +531,6 @@
deps = [
":audio_processing",
":audioproc_test_utils",
- "..:module_api",
"../..:webrtc_common",
"../../base:gtest_prod",
"../../base:protobuf_utils",
@@ -751,7 +749,6 @@
deps = [
":audio_processing",
- "..:module_api",
"../../base:rtc_base_approved",
"../../common_audio",
"../../system_wrappers:system_wrappers",
@@ -767,7 +764,6 @@
]
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 3dcbb27..647079a 100644
--- a/webrtc/modules/congestion_controller/BUILD.gn
+++ b/webrtc/modules/congestion_controller/BUILD.gn
@@ -45,7 +45,6 @@
}
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 7ab897f..4f8fbbc 100644
--- a/webrtc/modules/media_file/BUILD.gn
+++ b/webrtc/modules/media_file/BUILD.gn
@@ -33,7 +33,6 @@
}
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 57126d7..ce2356e 100644
--- a/webrtc/modules/pacing/BUILD.gn
+++ b/webrtc/modules/pacing/BUILD.gn
@@ -26,7 +26,6 @@
}
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 04f2f7c..c2e5d31 100644
--- a/webrtc/modules/remote_bitrate_estimator/BUILD.gn
+++ b/webrtc/modules/remote_bitrate_estimator/BUILD.gn
@@ -109,7 +109,6 @@
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 5d754d9..a369218 100644
--- a/webrtc/modules/rtp_rtcp/BUILD.gn
+++ b/webrtc/modules/rtp_rtcp/BUILD.gn
@@ -166,7 +166,6 @@
}
deps = [
- "..:module_api",
"../..:webrtc_common",
"../../api:libjingle_peerconnection_api",
"../../api:transport_api",
@@ -201,7 +200,6 @@
]
deps = [
":rtp_rtcp",
- "..:module_api",
"../../base:rtc_base_approved",
]
@@ -259,7 +257,6 @@
]
deps = [
":rtp_rtcp",
- "..:module_api",
"../../base:rtc_base_approved",
"../../test:test_support",
]
@@ -339,7 +336,6 @@
":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 7123890..3d32ac2 100644
--- a/webrtc/modules/utility/BUILD.gn
+++ b/webrtc/modules/utility/BUILD.gn
@@ -30,7 +30,6 @@
}
deps = [
- "..:module_api",
"../..:webrtc_common",
"../../audio/utility:audio_frame_operations",
"../../base:rtc_task_queue",
@@ -55,7 +54,6 @@
]
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 c902ee8..b7482a2 100644
--- a/webrtc/modules/video_capture/BUILD.gn
+++ b/webrtc/modules/video_capture/BUILD.gn
@@ -26,7 +26,6 @@
]
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 477e064..bafc8ba 100644
--- a/webrtc/modules/video_coding/BUILD.gn
+++ b/webrtc/modules/video_coding/BUILD.gn
@@ -94,7 +94,6 @@
":webrtc_i420",
":webrtc_vp8",
":webrtc_vp9",
- "..:module_api",
"../..:video_stream_api",
"../..:webrtc_common",
"../../base:rtc_base",
@@ -130,7 +129,6 @@
}
deps = [
- "..:module_api",
"../..:webrtc_common",
"../../api/video_codecs:video_codecs_api",
"../../base:rtc_base_approved",
@@ -227,7 +225,6 @@
deps = [
":video_coding_utility",
- "..:module_api",
"../..:webrtc_common",
"../../api/video_codecs:video_codecs_api",
"../../base:rtc_base_approved",
@@ -263,7 +260,6 @@
deps = [
":video_coding_utility",
- "..:module_api",
"../../base:rtc_base_approved",
"../../common_video",
"../../system_wrappers",
@@ -547,7 +543,6 @@
":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 c4c9c3b..7c9391a 100644
--- a/webrtc/modules/video_processing/BUILD.gn
+++ b/webrtc/modules/video_processing/BUILD.gn
@@ -26,7 +26,6 @@
deps = [
":denoiser_filter",
- "..:module_api",
"../../base:rtc_base_approved",
"../../common_audio",
"../../common_video",
@@ -52,9 +51,6 @@
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 ca6054e..a352f9c 100644
--- a/webrtc/sdk/BUILD.gn
+++ b/webrtc/sdk/BUILD.gn
@@ -423,7 +423,6 @@
"../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 2850552..be3296d 100644
--- a/webrtc/tools/BUILD.gn
+++ b/webrtc/tools/BUILD.gn
@@ -210,7 +210,6 @@
"../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 @@
}
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 61d628a..76f6ece 100644
--- a/webrtc/video/BUILD.gn
+++ b/webrtc/video/BUILD.gn
@@ -65,7 +65,6 @@
"../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 @@
"../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 be6faab..ca774f2 100644
--- a/webrtc/voice_engine/BUILD.gn
+++ b/webrtc/voice_engine/BUILD.gn
@@ -16,7 +16,6 @@
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 @@
"..:webrtc_common",
"../base:rtc_base_approved",
"../common_audio",
- "../modules:module_api",
"../modules/media_file",
]
@@ -60,7 +58,6 @@
"..:webrtc_common",
"../base:rtc_base_approved",
"../common_audio",
- "../modules:module_api",
"../modules/media_file:media_file",
"../system_wrappers",
]
@@ -144,7 +141,6 @@
"../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 @@
"..:webrtc_common",
"../base:rtc_base_approved",
"../common_audio",
- "../modules:module_api",
]
}
@@ -186,7 +181,6 @@
":file_player",
":voice_engine",
"../base:rtc_base_approved",
- "../modules:module_api",
"../test:test_common",
"//testing/gmock",
"//testing/gtest",
@@ -250,7 +244,6 @@
":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",