Bug 1376873 - Update build files from tip of webrtc.org; r=ng

This is a cherrypick and rebase of the build system changes in commit
9c6861308021b1f05fbf17538778ee7177e6d312. These landed on webrtc.org after 64
was branched.

MozReview-Commit-ID: JQKeXWdlkq6

Differential Revision: https://phabricator.services.mozilla.com/D7422

--HG--
extra : rebase_source : b66aef30518f8d26bb8dc8a62fcb504119479c85

media/webrtc/trunk/webrtc/BUILD.gn

@@ -9,7 +9,9 @@
 import("//build/config/linux/pkg_config.gni")
 import("//build/config/sanitizers/sanitizers.gni")
 import("webrtc.gni")
-import("//third_party/protobuf/proto_library.gni")
+if (!build_with_mozilla) {
+  import("//third_party/protobuf/proto_library.gni")
+}
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")
@@ -94,6 +96,10 @@ config("common_inherited_config") {
       "WEBRTC_LINUX",
       "WEBRTC_ANDROID",
     ]
+
+    if (build_with_mozilla) {
+      defines += [ "WEBRTC_ANDROID_OPENSLES" ]
+    }
   }
   if (is_chromeos) {
     defines += [ "CHROMEOS" ]
@@ -268,27 +274,38 @@ if (!build_with_chromium) {
 
     deps = [
       ":webrtc_common",
-      "api",
       "api:transport_api",
       "audio",
       "call",
       "common_audio",
       "common_video",
-      "logging",
       "media",
       "modules",
       "modules/video_capture:video_capture_internal_impl",
       "ortc",
-      "p2p",
-      "pc",
       "rtc_base",
       "sdk",
-      "stats",
       "system_wrappers:system_wrappers_default",
       "video",
       "voice_engine",
     ]
 
+    if (build_with_mozilla) {
+      deps += [
+        "api:video_frame_api",
+        "system_wrappers:field_trial_default",
+        "system_wrappers:metrics_default",
+      ]
+    } else {
+      deps += [
+        "api",
+        "logging",
+        "p2p",
+        "pc",
+        "stats",
+      ]
+    }
+
     if (rtc_enable_protobuf) {
       defines += [ "ENABLE_RTC_EVENT_LOG" ]
       deps += [ "logging:rtc_event_log_proto" ]

media/webrtc/trunk/webrtc/api/BUILD.gn

@@ -13,9 +13,11 @@ if (is_android) {
 }
 
 group("api") {
-  public_deps = [
-    ":libjingle_peerconnection_api",
-  ]
+  public_deps = []
+
+  if (!build_with_mozilla) {
+    public_deps += [ ":libjingle_peerconnection_api" ]
+  }
 }
 
 rtc_source_set("call_api") {
@@ -336,7 +338,6 @@ if (rtc_include_tests) {
       "fakemetricsobserver.h",
     ]
     deps = [
-      ":libjingle_peerconnection_api",
       "../api:peerconnection_and_implicit_call_api",
       "../media:rtc_media_base",
       "../rtc_base:rtc_base_approved",
@@ -345,6 +346,9 @@ if (rtc_include_tests) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
+    if (!build_with_mozilla) {
+      deps += [ ":libjingle_peerconnection_api" ]
+    }
   }
 
   rtc_source_set("rtc_api_unittests") {

media/webrtc/trunk/webrtc/call/BUILD.gn

@@ -11,7 +11,6 @@ import("../webrtc.gni")
 rtc_source_set("call_interfaces") {
   sources = [
     "audio_receive_stream.h",
-    "audio_send_stream.cc",
     "audio_send_stream.h",
     "audio_state.h",
     "call.h",
@@ -20,6 +19,9 @@ rtc_source_set("call_interfaces") {
     "syncable.cc",
     "syncable.h",
   ]
+  if (!build_with_mozilla) {
+    sources += [ "audio_send_stream.cc" ]
+  }
   deps = [
     ":rtp_interfaces",
     ":video_stream_api",

media/webrtc/trunk/webrtc/logging/BUILD.gn

@@ -7,7 +7,9 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 import("../webrtc.gni")
-import("//third_party/protobuf/proto_library.gni")
+if (!build_with_mozilla) {
+  import("//third_party/protobuf/proto_library.gni")
+}
 if (is_android) {
   import("//build/config/android/config.gni")
   import("//build/config/android/rules.gni")

media/webrtc/trunk/webrtc/media/BUILD.gn

@@ -10,10 +10,13 @@ import("//build/config/linux/pkg_config.gni")
 import("../webrtc.gni")
 
 group("media") {
-  public_deps = [
-    ":rtc_media",
-    ":rtc_media_base",
-  ]
+  public_deps = []
+  if (!build_with_mozilla) {
+    public_deps += [
+      ":rtc_media",
+      ":rtc_media_base",
+    ]
+  }
 }
 
 config("rtc_media_defines_config") {
@@ -120,7 +123,6 @@ rtc_static_library("rtc_media_base") {
     "../call:call_interfaces",
     "../call:video_stream_api",
     "../modules/audio_processing:audio_processing_statistics",
-    "../p2p",
     "../rtc_base:rtc_base",
     "../rtc_base:rtc_base_approved",
     "../system_wrappers:field_trial_api",
@@ -128,6 +130,10 @@ rtc_static_library("rtc_media_base") {
 
   public_deps += [ ":rtc_h264_profile_id" ]
 
+  if (!build_with_mozilla) {
+    deps += [ "../p2p" ]
+  }
+
   if (is_nacl) {
     deps += [ "//native_client_sdk/src/libraries/nacl_io" ]
   }

media/webrtc/trunk/webrtc/modules/BUILD.gn

@@ -255,11 +255,14 @@ if (rtc_include_tests) {
       "remote_bitrate_estimator:remote_bitrate_estimator_unittests",
       "rtp_rtcp:rtp_rtcp_unittests",
       "utility:utility_unittests",
-      "video_capture",
       "video_coding:video_coding_unittests",
       "video_processing:video_processing_unittests",
     ]
 
+    if (!build_with_mozilla) {
+      deps += [ "video_capture" ]
+    }
+
     data = modules_unittests_resources
 
     if (is_android) {

media/webrtc/trunk/webrtc/modules/audio_coding/BUILD.gn

@@ -9,7 +9,9 @@
 import("../../webrtc.gni")
 import("audio_coding.gni")
 import("//build/config/arm.gni")
-import("//third_party/protobuf/proto_library.gni")
+if (!build_with_mozilla) {
+  import("//third_party/protobuf/proto_library.gni")
+}
 
 audio_codec_deps = [
   ":cng",
@@ -808,7 +810,7 @@ rtc_static_library("webrtc_opus") {
   if (rtc_build_opus) {
     public_deps += [ rtc_opus_dir ]
   } else if (build_with_mozilla) {
-    include_dirs = [ getenv("DIST") + "/include/opus" ]
+    include_dirs = [ "/media/libopus/include" ]
   }
 }
 

media/webrtc/trunk/webrtc/modules/audio_device/BUILD.gn

@@ -123,6 +123,13 @@ rtc_source_set("audio_device_generic") {
     "include/audio_device_defines.h",
   ]
 
+  if (build_with_mozilla) {
+    sources += [
+      "opensl/single_rw_fifo.cc",
+      "opensl/single_rw_fifo.h",
+    ]
+  }
+
   include_dirs = []
   if (is_linux) {
     include_dirs += [ "linux" ]
@@ -173,6 +180,14 @@ rtc_source_set("audio_device_generic") {
         "log",
         "OpenSLES",
       ]
+
+      if (build_with_mozilla) {
+        include_dirs += [
+          "/config/external/nspr",
+          "/nsprpub/lib/ds",
+          "/nsprpub/pr/include",
+        ]
+      }
     }
     if (rtc_use_dummy_audio_file_devices) {
       defines += [ "WEBRTC_DUMMY_FILE_DEVICES" ]

media/webrtc/trunk/webrtc/modules/audio_processing/BUILD.gn

@@ -7,8 +7,10 @@
 # be found in the AUTHORS file in the root of the source tree.
 
 import("//build/config/arm.gni")
-import("//third_party/protobuf/proto_library.gni")
 import("../../webrtc.gni")
+if (!build_with_mozilla) {
+  import("//third_party/protobuf/proto_library.gni")
+}
 
 declare_args() {
   # Disables the usual mode where we trust the reported system delay

media/webrtc/trunk/webrtc/modules/congestion_controller/BUILD.gn

@@ -61,6 +61,10 @@ rtc_static_library("congestion_controller") {
     "../rtp_rtcp",
     "../utility",
   ]
+
+  if (!build_with_mozilla) {
+    deps += [ "../../rtc_base:rtc_base" ]
+  }
 }
 
 if (rtc_include_tests) {

media/webrtc/trunk/webrtc/modules/desktop_capture/BUILD.gn

@@ -28,8 +28,11 @@ rtc_static_library("primitives") {
 
   deps = [
     "../..:webrtc_common",
-    "../../rtc_base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
   ]
+
+  if (!build_with_mozilla) {
+    deps += [ "../../rtc_base:rtc_base" ]  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
+  }
 }
 
 if (rtc_include_tests) {
@@ -287,6 +290,16 @@ rtc_static_library("desktop_capture_generic") {
     "window_finder_win.h",
   ]
 
+  if (build_with_mozilla) {
+    sources += [
+      "app_capturer_win.cc",
+      "desktop_device_info.cc",
+      "desktop_device_info.h",
+      "win/desktop_device_info_win.cc",
+      "win/win_shared.cc",
+    ]
+  }
+
   if (use_x11) {
     sources += [
       "mouse_cursor_monitor_x11.cc",
@@ -306,6 +319,17 @@ rtc_static_library("desktop_capture_generic") {
       "x11/x_server_pixel_buffer.h",
     ]
     configs += [ "//build/config/linux:x11" ]
+
+    if (build_with_mozilla) {
+      sources += [
+        "app_capturer_x11.cc",
+        "app_capturer_x11.h",
+        "x11/desktop_device_info_x11.cc",
+        "x11/desktop_device_info_x11.h",
+        "x11/shared_x_util.cc",
+        "x11/shared_x_util.h",
+      ]
+    }
   }
 
   if (!is_win && !is_mac && !use_x11) {
@@ -329,9 +353,14 @@ rtc_static_library("desktop_capture_generic") {
     "../../api:refcountedbase",
     "../../rtc_base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
     "../../system_wrappers",
-    "//third_party/libyuv",
   ]
 
+  if (build_with_mozilla) {
+    deps += [ "../../rtc_base:rtc_base_approved" ]
+  } else {
+    deps += [ "//third_party/libyuv" ]
+  }
+
   if (use_desktop_capture_differ_sse2) {
     deps += [ ":desktop_capture_differ_sse2" ]
   }

media/webrtc/trunk/webrtc/modules/video_capture/BUILD.gn

@@ -158,7 +158,16 @@ if (!build_with_chromium) {
 
       libs = [ "Strmiids.lib" ]
 
-      deps += [ "//third_party/winsdk_samples" ]
+      if (build_with_mozilla) {
+        sources += [
+          "windows/BaseFilter.cpp",
+          "windows/BaseInputPin.cpp",
+          "windows/BasePin.cpp",
+          "windows/MediaType.cpp",
+        ]
+      } else {
+        deps += [ "//third_party/winsdk_samples" ]
+      }
     }
     if (is_ios || is_mac) {
       deps += [ ":video_capture_internal_impl_objc" ]
@@ -170,6 +179,19 @@ if (!build_with_chromium) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
+
+    if (build_with_mozilla && is_android) {
+      include_dirs = [
+        "/config/external/nspr",
+        "/nsprpub/lib/ds",
+        "/nsprpub/pr/include",
+      ]
+
+      sources = [
+        "android/device_info_android.cc",
+        "android/video_capture_android.cc",
+      ]
+    }
   }
 
   if (!is_android && rtc_include_tests) {

media/webrtc/trunk/webrtc/modules/video_coding/BUILD.gn

@@ -211,7 +211,6 @@ rtc_static_library("webrtc_h264") {
     ]
     deps += [
       "../../common_video",
-      "../../media:rtc_media_base",
       "//third_party/ffmpeg:ffmpeg",
       "//third_party/openh264:encoder",
     ]
@@ -363,6 +362,10 @@ if (rtc_include_tests) {
         suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
       }
     }
+
+    if (!build_with_mozilla) {
+      deps += [ "../../media:rtc_media_base" ]
+    }
   }
 
   rtc_source_set("simulcast_test_utility") {

media/webrtc/trunk/webrtc/rtc_base/BUILD.gn

@@ -55,7 +55,7 @@ config("rtc_base_all_dependent_config") {
   }
 }
 
-if (!rtc_build_ssl) {
+if (!rtc_build_ssl && !build_with_mozilla) {
   config("external_ssl_library") {
     assert(rtc_ssl_root != "",
            "You must specify rtc_ssl_root when rtc_build_ssl==0.")
@@ -445,9 +445,11 @@ rtc_source_set("rtc_json") {
 }
 
 rtc_static_library("rtc_base") {
-  public_deps = [
-    ":rtc_base_generic",
-  ]
+  public_deps = []
+  if (!build_with_mozilla) {
+    public_deps += [ ":rtc_base_generic" ]
+  }
+
   if (is_win) {
     sources = [
       "noop.cc",
@@ -634,7 +636,7 @@ rtc_static_library("rtc_base_generic") {
 
   if (rtc_build_ssl) {
     deps += [ "//third_party/boringssl" ]
-  } else {
+  } else if (!build_with_mozilla) {
     configs += [ ":external_ssl_library" ]
   }
 

media/webrtc/trunk/webrtc/system_wrappers/BUILD.gn

@@ -63,7 +63,15 @@ rtc_static_library("system_wrappers") {
   if (is_android) {
     defines += [ "WEBRTC_THREAD_RR" ]
 
-    deps += [ ":cpu_features_android" ]
+    if (build_with_mozilla) {
+      include_dirs = [
+        "/config/external/nspr",
+        "/nsprpub/lib/ds",
+        "/nsprpub/pr/include",
+      ]
+    } else {
+      deps += [ ":cpu_features_android" ]
+    }
 
     libs += [ "log" ]
   }
@@ -167,7 +175,7 @@ group("system_wrappers_default") {
   ]
 }
 
-if (is_android) {
+if (is_android && !build_with_mozilla) {
   rtc_static_library("cpu_features_android") {
     sources = [
       "source/cpu_features_android.c",

media/webrtc/trunk/webrtc/video/BUILD.gn

@@ -68,7 +68,6 @@ rtc_static_library("video") {
     "../call:rtp_receiver",
     "../common_video",
     "../logging:rtc_event_log_api",
-    "../media:rtc_media_base",
     "../modules:module_api",
     "../modules/bitrate_controller",
     "../modules/congestion_controller",
@@ -88,6 +87,10 @@ rtc_static_library("video") {
     "../system_wrappers",
     "../voice_engine",
   ]
+
+  if (!build_with_mozilla) {
+    deps += [ "../media:rtc_media_base" ]
+  }
 }
 
 if (rtc_include_tests) {
@@ -307,5 +310,8 @@ if (rtc_include_tests) {
     if (rtc_use_h264) {
       defines += [ "WEBRTC_USE_H264" ]
     }
+    if (!build_with_mozilla) {
+      deps += [ "../media:rtc_media_base" ]
+    }
   }
 }

media/webrtc/trunk/webrtc/webrtc.gni

@@ -5,14 +5,12 @@
 # tree. An additional intellectual property rights grant can be found
 # in the file PATENTS.  All contributing project authors may
 # be found in the AUTHORS file in the root of the source tree.
-
 import("//build/config/arm.gni")
 import("//build/config/features.gni")
 import("//build/config/mips.gni")
 import("//build/config/sanitizers/sanitizers.gni")
 import("//build/config/ui.gni")
 import("//build_overrides/build.gni")
-import("//testing/test.gni")
 
 if (!build_with_chromium && is_component_build) {
   print("The Gn argument `is_component_build` is currently " +
@@ -57,9 +55,6 @@ declare_args() {
   # Selects fixed-point code where possible.
   rtc_prefer_fixed_point = false
 
-  # Enables the use of protocol buffers for debug recordings.
-  rtc_enable_protobuf = true
-
   # Disable the code for the intelligibility enhancer by default.
   rtc_enable_intelligibility_enhancer = false
 
@@ -74,20 +69,6 @@ declare_args() {
   # Set this to true to enable BWE test logging.
   rtc_enable_bwe_test_logging = false
 
-  # Set this to disable building with support for SCTP data channels.
-  rtc_enable_sctp = true
-
-  # Disable these to not build components which can be externally provided.
-  rtc_build_json = true
-  rtc_build_libsrtp = true
-  rtc_build_libvpx = true
-  rtc_libvpx_build_vp9 = true
-  rtc_build_libyuv = true
-  rtc_build_openmax_dl = true
-  rtc_build_opus = true
-  rtc_build_ssl = true
-  rtc_build_usrsctp = true
-
   # Enable to use the Mozilla internal settings.
   build_with_mozilla = false
 
@@ -110,28 +91,10 @@ declare_args() {
   # use an external implementation.
   rtc_link_task_queue_impl = true
 
-  # Enable libevent task queues on platforms that support it.
-  # rtc_link_task_queue_impl must be set to true for this to
-  # have an effect.
-  if (is_win || is_mac || is_ios || is_nacl) {
-    rtc_enable_libevent = false
-    rtc_build_libevent = false
-  } else {
-    rtc_enable_libevent = true
-    rtc_build_libevent = true
-  }
-
   if (current_cpu == "arm" || current_cpu == "arm64") {
     rtc_prefer_fixed_point = true
   }
 
-  if (!is_ios && (current_cpu != "arm" || arm_version >= 7) &&
-      current_cpu != "mips64el") {
-    rtc_use_openmax_dl = true
-  } else {
-    rtc_use_openmax_dl = false
-  }
-
   # Determines whether NEON code will be built.
   rtc_build_with_neon =
       (current_cpu == "arm" && arm_use_neon) || current_cpu == "arm64"
@@ -166,17 +129,55 @@ declare_args() {
   # as Chromium, must turn this flag off so that WebRTC does not also
   # initialize.
   rtc_initialize_ffmpeg = !build_with_chromium
+}
 
-  # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
-  # build environments, even if available for Chromium builds.
-  rtc_use_gtk = !build_with_chromium
+if (!build_with_mozilla) {
+  import("//testing/test.gni")
 }
 
 # A second declare_args block, so that declarations within it can
 # depend on the possibly overridden variables in the first
 # declare_args block.
 declare_args() {
-  rtc_restrict_logging = build_with_chromium
+  # Enables the use of protocol buffers for debug recordings.
+  rtc_enable_protobuf = !build_with_mozilla
+
+  # Set this to disable building with support for SCTP data channels.
+  rtc_enable_sctp = !build_with_mozilla
+
+  # Disable these to not build components which can be externally provided.
+  rtc_build_json = !build_with_mozilla
+  rtc_build_libsrtp = !build_with_mozilla
+  rtc_build_libvpx = !build_with_mozilla
+  rtc_libvpx_build_vp9 = !build_with_mozilla
+  rtc_build_openmax_dl = !build_with_mozilla
+  rtc_build_opus = !build_with_mozilla
+  rtc_build_ssl = !build_with_mozilla
+  rtc_build_usrsctp = !build_with_mozilla
+
+  # Enable libevent task queues on platforms that support it.
+  # rtc_link_task_queue_impl must be set to true for this to
+  # have an effect.
+  if (is_win || is_mac || is_ios || is_nacl) {
+    rtc_enable_libevent = false
+    rtc_build_libevent = false
+  } else {
+    rtc_enable_libevent = true
+    rtc_build_libevent = !build_with_mozilla
+  }
+
+  if (!is_ios && (current_cpu != "arm" || arm_version >= 7) &&
+      current_cpu != "mips64el" && !build_with_mozilla) {
+    rtc_use_openmax_dl = true
+  } else {
+    rtc_use_openmax_dl = false
+  }
+
+  # Build sources requiring GTK. NOTICE: This is not present in Chrome OS
+  # build environments, even if available for Chromium builds.
+  rtc_use_gtk = !build_with_chromium && !build_with_mozilla
+
+  rtc_restrict_logging = build_with_chromium || build_with_mozilla
 
   # Excluded in Chromium since its prerequisites don't require Pulse Audio.
   rtc_include_pulse_audio = !build_with_chromium
@@ -186,7 +187,7 @@ declare_args() {
   rtc_include_internal_audio_device = !build_with_chromium
 
   # Include tests in standalone checkout.
-  rtc_include_tests = !build_with_chromium
+  rtc_include_tests = !build_with_chromium && !build_with_mozilla
 }
 
 # Make it possible to provide custom locations for some libraries (move these