Reland of Creating webrtc/modules:module_api (patchset #1 id:1 of https://codereview.webrtc.org/2839963005/ )

Reason for revert:
Fixing the Gn error and try to reland.

Original issue's description:
> 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: 5a1a092ed0
>
> 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}
> Committed: bb08c3e296

TBR=kjellander@webrtc.org
# Skipping CQ checks because original CL landed less than 1 days ago.
NOPRESUBMIT=true
NOTREECHECKS=true
NOTRY=true
TBR=kjellander@webrtc.org
BUG=webrtc:7513

Review-Url: https://codereview.webrtc.org/2843913002
Cr-Commit-Position: refs/heads/master@{#17884}

webrtc/api/BUILD.gn

@@ -156,6 +156,7 @@ rtc_source_set("audio_mixer_api") {
 
   deps = [
     "../base:rtc_base_approved",
+    "../modules:module_api",
   ]
 }
 

webrtc/audio/utility/BUILD.gn

@@ -22,6 +22,7 @@ rtc_static_library("audio_frame_operations") {
   deps = [
     "../..:webrtc_common",
     "../../base:rtc_base_approved",
+    "../../modules:module_api",
     "../../modules/audio_coding:audio_format_conversion",
   ]
 }
@@ -35,6 +36,7 @@ if (rtc_include_tests) {
     deps = [
       ":audio_frame_operations",
       "../../base:rtc_base_approved",
+      "../../modules:module_api",
       "../../test:test_support",
       "//testing/gtest",
     ]

webrtc/common_video/BUILD.gn

@@ -59,6 +59,7 @@ rtc_static_library("common_video") {
     "..:webrtc_common",
     "../base:rtc_base",
     "../base:rtc_task_queue",
+    "../modules:module_api",
     "../system_wrappers",
   ]
   public_deps = [

webrtc/modules/BUILD.gn

@@ -29,6 +29,18 @@ 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",
@@ -199,8 +211,6 @@ if (rtc_include_tests) {
 
   rtc_test("modules_unittests") {
     testonly = true
-
-    deps = []
     defines = []
     sources = [
       "module_common_types_unittest.cc",
@@ -211,7 +221,8 @@ if (rtc_include_tests) {
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
-    deps += [
+    deps = [
+      ":module_api",
       "../test:test_main",
       "audio_coding:audio_coding_unittests",
       "audio_conference_mixer:audio_conference_mixer_unittests",

webrtc/modules/audio_coding/BUILD.gn

@@ -127,6 +127,7 @@ rtc_source_set("audio_coding_module_typedefs") {
     "include/audio_coding_module_typedefs.h",
   ]
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
   ]
 }
@@ -163,6 +164,7 @@ 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",
@@ -1067,6 +1069,7 @@ rtc_static_library("neteq") {
     ":isac_fix",
     ":neteq_decoder_enum",
     ":pcm16b",
+    "..:module_api",
     "../..:webrtc_common",
     "../../api/audio_codecs:audio_codecs_api",
     "../../base:gtest_prod",
@@ -1198,6 +1201,7 @@ 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",
@@ -1301,6 +1305,7 @@ if (rtc_include_tests) {
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":audio_format_conversion",
+      "..:module_api",
       "../../:webrtc_common",
       "../../base:rtc_base_approved",
       "../../system_wrappers",
@@ -1328,6 +1333,7 @@ if (rtc_include_tests) {
     deps = [
       ":audio_coding",
       ":audio_format_conversion",
+      "..:module_api",
       "../../:webrtc_common",
       "../../base:rtc_base_approved",
       "../../system_wrappers",
@@ -1429,7 +1435,9 @@ if (rtc_include_tests) {
     rtc_test("neteq_rtpplay") {
       testonly = true
       defines = []
-      deps = []
+      deps = [
+        "..:module_api",
+      ]
       sources = [
         "neteq/tools/neteq_rtpplay.cc",
       ]
@@ -1509,6 +1517,7 @@ 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",
@@ -1534,6 +1543,7 @@ if (rtc_include_tests) {
     deps = [
       ":neteq",
       ":neteq_unittest_tools",
+      "..:module_api",
       "../..:webrtc_common",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../base:rtc_base_approved",
@@ -1588,6 +1598,7 @@ if (rtc_include_tests) {
     deps = [
       ":audio_encoder_interface",
       ":pcm16b",
+      "..:module_api",
       "../..:webrtc_common",
       "../../api/audio_codecs:audio_codecs_api",
       "../../base:rtc_base_approved",
@@ -1625,6 +1636,7 @@ if (rtc_include_tests) {
       ":ilbc",
       ":isac",
       ":pcm16b",
+      "..:module_api",
       "../..:webrtc_common",
       "//testing/gtest",
     ]
@@ -2136,6 +2148,7 @@ 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",

webrtc/modules/audio_conference_mixer/BUILD.gn

@@ -39,6 +39,7 @@ rtc_static_library("audio_conference_mixer") {
   }
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../audio/utility:audio_frame_operations",
     "../../base:rtc_base_approved",

webrtc/modules/audio_device/BUILD.gn

@@ -49,6 +49,7 @@ rtc_static_library("audio_device") {
   public_configs = [ ":audio_device_config" ]
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../base:rtc_base_approved",
     "../../base:rtc_task_queue",

webrtc/modules/audio_mixer/BUILD.gn

@@ -38,6 +38,7 @@ rtc_static_library("audio_mixer_impl") {
 
   deps = [
     ":audio_frame_manipulator",
+    "..:module_api",
     "../..:webrtc_common",
     "../../audio/utility:audio_frame_operations",
     "../../base:rtc_base_approved",
@@ -58,6 +59,7 @@ rtc_static_library("audio_frame_manipulator") {
   ]
 
   deps = [
+    "..:module_api",
     "../../audio/utility",
     "../../base:rtc_base_approved",
   ]
@@ -85,6 +87,7 @@ 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",

webrtc/modules/audio_processing/BUILD.gn

@@ -230,6 +230,7 @@ rtc_static_library("audio_processing") {
 
   defines = []
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../audio/utility:audio_frame_operations",
     "../../base:gtest_prod",
@@ -531,6 +532,7 @@ if (rtc_include_tests) {
     deps = [
       ":audio_processing",
       ":audioproc_test_utils",
+      "..:module_api",
       "../..:webrtc_common",
       "../../base:gtest_prod",
       "../../base:protobuf_utils",
@@ -749,6 +751,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":audio_processing",
+      "..:module_api",
       "../../base:rtc_base_approved",
       "../../common_audio",
       "../../system_wrappers:system_wrappers",
@@ -764,6 +767,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":audio_processing",
+      "..:module_api",
       "../..:webrtc_common",
       "../../common_audio:common_audio",
       "../../system_wrappers:metrics_default",

webrtc/modules/congestion_controller/BUILD.gn

@@ -45,6 +45,7 @@ rtc_static_library("congestion_controller") {
   }
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../base:rtc_base",
     "../../base:rtc_base_approved",

webrtc/modules/media_file/BUILD.gn

@@ -33,6 +33,7 @@ rtc_static_library("media_file") {
   }
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../base:rtc_base_approved",
     "../../common_audio",

webrtc/modules/pacing/BUILD.gn

@@ -26,6 +26,7 @@ rtc_static_library("pacing") {
   }
 
   deps = [
+    "..:module_api",
     "../../:webrtc_common",
     "../../base:rtc_base_approved",
     "../../logging:rtc_event_log_api",

webrtc/modules/remote_bitrate_estimator/BUILD.gn

@@ -109,6 +109,7 @@ if (rtc_include_tests) {
 
     deps = [
       ":remote_bitrate_estimator",
+      "..:module_api",
       "../..:webrtc_common",
       "../../base:gtest_prod",
       "../../base:rtc_base",

webrtc/modules/rtp_rtcp/BUILD.gn

@@ -166,6 +166,7 @@ rtc_static_library("rtp_rtcp") {
   }
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../api:libjingle_peerconnection_api",
     "../../api:transport_api",
@@ -200,6 +201,7 @@ rtc_source_set("fec_test_helper") {
   ]
   deps = [
     ":rtp_rtcp",
+    "..:module_api",
     "../../base:rtc_base_approved",
   ]
 
@@ -257,6 +259,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":rtp_rtcp",
+      "..:module_api",
       "../../base:rtc_base_approved",
       "../../test:test_support",
     ]
@@ -336,6 +339,7 @@ if (rtc_include_tests) {
       ":fec_test_helper",
       ":mock_rtp_rtcp",
       ":rtp_rtcp",
+      "..:module_api",
       "../..:webrtc_common",
       "../../api:transport_api",
       "../../base:rtc_base_approved",

webrtc/modules/utility/BUILD.gn

@@ -30,6 +30,7 @@ rtc_static_library("utility") {
   }
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../audio/utility:audio_frame_operations",
     "../../base:rtc_task_queue",
@@ -54,6 +55,7 @@ if (rtc_include_tests) {
     ]
     deps = [
       ":utility",
+      "..:module_api",
       "../../base:rtc_task_queue",
       "../../test:test_support",
       "//testing/gmock",

webrtc/modules/video_capture/BUILD.gn

@@ -26,6 +26,7 @@ rtc_static_library("video_capture_module") {
   ]
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../base:rtc_base_approved",
     "../../common_video",

webrtc/modules/video_coding/BUILD.gn

@@ -94,6 +94,7 @@ rtc_static_library("video_coding") {
     ":webrtc_i420",
     ":webrtc_vp8",
     ":webrtc_vp9",
+    "..:module_api",
     "../..:video_stream_api",
     "../..:webrtc_common",
     "../../base:rtc_base",
@@ -129,6 +130,7 @@ rtc_static_library("video_coding_utility") {
   }
 
   deps = [
+    "..:module_api",
     "../..:webrtc_common",
     "../../api/video_codecs:video_codecs_api",
     "../../base:rtc_base_approved",
@@ -225,6 +227,7 @@ rtc_static_library("webrtc_vp8") {
 
   deps = [
     ":video_coding_utility",
+    "..:module_api",
     "../..:webrtc_common",
     "../../api/video_codecs:video_codecs_api",
     "../../base:rtc_base_approved",
@@ -260,6 +263,7 @@ rtc_static_library("webrtc_vp9") {
 
   deps = [
     ":video_coding_utility",
+    "..:module_api",
     "../../base:rtc_base_approved",
     "../../common_video",
     "../../system_wrappers",
@@ -543,6 +547,7 @@ if (rtc_include_tests) {
       ":webrtc_h264",
       ":webrtc_vp8",
       ":webrtc_vp9",
+      "..:module_api",
       "../..:webrtc_common",
       "../../api:video_frame_api",
       "../../api/video_codecs:video_codecs_api",

webrtc/modules/video_processing/BUILD.gn

@@ -26,6 +26,7 @@ rtc_static_library("video_processing") {
 
   deps = [
     ":denoiser_filter",
+    "..:module_api",
     "../../base:rtc_base_approved",
     "../../common_audio",
     "../../common_video",
@@ -51,6 +52,9 @@ rtc_source_set("denoiser_filter") {
   sources = [
     "util/denoiser_filter.h",
   ]
+  deps = [
+    "..:module_api",
+  ]
 }
 
 if (build_video_processing_sse2) {

webrtc/sdk/BUILD.gn

@@ -425,6 +425,7 @@ 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",

webrtc/tools/BUILD.gn

@@ -210,6 +210,7 @@ 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.
@@ -261,6 +262,7 @@ if (rtc_include_tests) {
     }
 
     deps = [
+      "../modules:module_api",
       "../modules/audio_processing",
       "../system_wrappers:metrics_default",
       "../test:test_support",

webrtc/video/BUILD.gn

@@ -65,6 +65,7 @@ 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",
@@ -260,6 +261,7 @@ 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",

webrtc/voice_engine/BUILD.gn

@@ -16,6 +16,7 @@ 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",
@@ -39,6 +40,7 @@ rtc_static_library("file_player") {
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../common_audio",
+    "../modules:module_api",
     "../modules/media_file",
   ]
 
@@ -58,6 +60,7 @@ rtc_static_library("file_recorder") {
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../common_audio",
+    "../modules:module_api",
     "../modules/media_file:media_file",
     "../system_wrappers",
   ]
@@ -141,6 +144,7 @@ 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/.
@@ -172,6 +176,7 @@ rtc_static_library("audio_level") {
     "..:webrtc_common",
     "../base:rtc_base_approved",
     "../common_audio",
+    "../modules:module_api",
   ]
 }
 
@@ -181,6 +186,7 @@ if (rtc_include_tests) {
       ":file_player",
       ":voice_engine",
       "../base:rtc_base_approved",
+      "../modules:module_api",
       "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",
@@ -244,6 +250,7 @@ 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",