Removing unneeded webrtc dependencies on iOS.

BUILD.gn

 # Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
 #
 # Use of this source code is governed by a BSD-style license
 # that can be found in the LICENSE file in the root of the source
 # 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.
 
 # TODO(kjellander): Rebase this to webrtc/build/common.gypi changes after r6330.
 
@@ skipping 146 matching lines @@
     # The Android NDK doesn"t provide optimized versions of these
     # functions. Ensure they are disabled for all compilers.
     cflags += [
       "-fno-builtin-cos",
       "-fno-builtin-sin",
       "-fno-builtin-cosf",
       "-fno-builtin-sinf",
     ]
   }
 }
+if (!is_ios) {
+  source_set("webrtc") {
+    sources = [
+      "call.h",
+      "config.h",
+      "frame_callback.h",
+      "transport.h",
+    ]
 
-source_set("webrtc") {
-  sources = [
-    "call.h",
-    "config.h",
-    "frame_callback.h",
-    "transport.h",
-  ]
+    defines = []
+    configs += [ ":common_config" ]
+    public_configs = [ ":common_inherited_config" ]
 
-  defines = []
-  configs += [ ":common_config" ]
-  public_configs = [ ":common_inherited_config" ]
+    deps = [
+      ":webrtc_common",
+      "base:rtc_base",
+      "common_audio",
+      "common_video",
+      "modules/audio_coding",
+      "modules/audio_conference_mixer",
+      "modules/audio_device",
+      "modules/audio_processing",
+      "modules/bitrate_controller",
+      "modules/desktop_capture",
+      "modules/media_file",
+      "modules/rtp_rtcp",
+      "modules/utility",
+      "modules/video_coding",
+      "modules/video_processing",
+      "system_wrappers",
+      "tools",
+      "video",
+      "voice_engine",
+    ]
 
-  deps = [
-    ":webrtc_common",
-    "base:rtc_base",
-    "common_audio",
-    "common_video",
-    "modules/audio_coding",
-    "modules/audio_conference_mixer",
-    "modules/audio_device",
-    "modules/audio_processing",
-    "modules/bitrate_controller",
-    "modules/desktop_capture",
-    "modules/media_file",
-    "modules/rtp_rtcp",
-    "modules/utility",
-    "modules/video_coding",
-    "modules/video_processing",
-    "system_wrappers",
-    "tools",
-    "video",
-    "voice_engine",
-  ]
+    if (build_with_chromium) {
+      deps += [
+        "modules/video_capture",
+        "modules/video_render",
+      ]
+    }
 
-  if (build_with_chromium) {
-    deps += [
-      "modules/video_capture",
-      "modules/video_render",
-    ]
-  }
-
-  if (rtc_enable_protobuf) {
-    defines += [ "ENABLE_RTC_EVENT_LOG" ]
-    deps += [ ":rtc_event_log_proto" ]
+    if (rtc_enable_protobuf) {
+      defines += [ "ENABLE_RTC_EVENT_LOG" ]
+      deps += [ ":rtc_event_log_proto" ]
+    }
   }
 }
 
 if (!build_with_chromium) {
   executable("webrtc_tests") {
     testonly = true
     deps = [
       ":webrtc",
       "modules/video_render:video_render_internal_impl",
       "modules/video_capture:video_capture_internal_impl",
@@ skipping 15 matching lines @@
   configs += [ ":common_config" ]
   public_configs = [ ":common_inherited_config" ]
 }
 
 source_set("gtest_prod") {
   sources = [
     "test/testsupport/gtest_prod_util.h",
   ]
 }
 
-if (rtc_enable_protobuf) {
-  proto_library("rtc_event_log_proto") {
+if (!is_ios) {
+  if (rtc_enable_protobuf) {
+    proto_library("rtc_event_log_proto") {
+      sources = [
+        "video/rtc_event_log.proto",
+      ]
+      proto_out_dir = "webrtc/video"
+    }
+  }
+
+  source_set("rtc_event_log") {
     sources = [
-      "video/rtc_event_log.proto",
+      "video/rtc_event_log.cc",
+      "video/rtc_event_log.h",
     ]
-    proto_out_dir = "webrtc/video"
+
+    defines = []
+    configs += [ ":common_config" ]
+    public_configs = [ ":common_inherited_config" ]
+
+    deps = [
+      ":webrtc_common",
+    ]
+
+    if (rtc_enable_protobuf) {
+      defines += [ "ENABLE_RTC_EVENT_LOG" ]
+      deps += [ ":rtc_event_log_proto" ]
+    }
+    if (is_clang && !is_nacl) {
+      # Suppress warnings from Chrome's Clang plugins.
+      # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
+      configs -= [ "//build/config/clang:find_bad_constructs" ]
+    }
   }
 }
-
-source_set("rtc_event_log") {
-  sources = [
-    "video/rtc_event_log.cc",
-    "video/rtc_event_log.h",
-  ]
-
-  defines = []
-  configs += [ ":common_config" ]
-  public_configs = [ ":common_inherited_config" ]
-
-  deps = [
-    ":webrtc_common",
-  ]
-
-  if (rtc_enable_protobuf) {
-    defines += [ "ENABLE_RTC_EVENT_LOG" ]
-    deps += [ ":rtc_event_log_proto" ]
-  }
-  if (is_clang && !is_nacl) {
-    # Suppress warnings from Chrome's Clang plugins.
-    # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
-    configs -= [ "//build/config/clang:find_bad_constructs" ]
-  }
-}