Revert of Creating webrtc/modules:module_api (patchset #5 id:80001 of h…

…ttps://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 [email protected] # 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}
pristineio · Apr 26, 2017 · bb08c3e · bb08c3e
1 parent 5a1a092
commit bb08c3e
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Showing 22 changed files with 3 additions and 72 deletions.
diff --git 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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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
@@ -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",
-Original file line number
+Diff line change
@@ Expand Up / @@ -156,7 +156,6 @@ rtc_source_set("audio_mixer_api") { @@
       deps = [
         "../base:rtc_base_approved",
-        "../modules:module_api",
       ]
     }
@@ Expand Down @@