Move optional.h to webrtc/api/

webrtc/modules/audio_coding/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.
 
 import("../../webrtc.gni")
 import("audio_coding.gni")
@@ skipping 29 matching lines @@
                     ]
 
 rtc_static_library("audio_format_conversion") {
   sources = [
     "codecs/audio_format_conversion.cc",
     "codecs/audio_format_conversion.h",
   ]
   deps = [
     "../..:webrtc_common",
     "../../api:array_view",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("rent_a_codec") {
   sources = [
     "acm2/acm_codec_database.cc",
     "acm2/acm_codec_database.h",
     "acm2/rent_a_codec.cc",
     "acm2/rent_a_codec.h",
   ]
   deps = [
            "../../api:array_view",
+           "../../api:optional",
            "../../api/audio_codecs:audio_codecs_api",
            "../..:webrtc_common",
            "../../rtc_base:protobuf_utils",
            "../../rtc_base:rtc_base_approved",
            "../../system_wrappers",
            ":audio_coding_module_typedefs",
            ":isac_common",
            ":isac_fix_c",
            ":neteq_decoder_enum",
          ] + audio_codec_deps
@@ skipping 51 matching lines @@
 
   deps = audio_coding_deps + [
            "..:module_api",
            "../../api:array_view",
            "../../api/audio_codecs:audio_codecs_api",
            "../../api/audio_codecs:builtin_audio_decoder_factory",
            ":audio_coding_module_typedefs",
            ":neteq",
            ":rent_a_codec",
            "../../rtc_base:rtc_base_approved",
+           "../../api:optional",
            "../../logging:rtc_event_log_api",
          ]
   defines = audio_coding_defines
 }
 
 rtc_static_library("legacy_encoded_audio_frame") {
   sources = [
     "codecs/legacy_encoded_audio_frame.cc",
     "codecs/legacy_encoded_audio_frame.h",
   ]
@@ skipping 662 matching lines @@
   sources = [
     "codecs/opus/audio_decoder_opus.cc",
     "codecs/opus/audio_decoder_opus.h",
     "codecs/opus/audio_encoder_opus.cc",
     "codecs/opus/audio_encoder_opus.h",
   ]
 
   deps = [
     ":audio_network_adaptor",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs/opus:audio_encoder_opus_config",
     "../../common_audio",
     "../../rtc_base:rtc_base_approved",
     "../../rtc_base:rtc_numerics",
     "../../system_wrappers",
   ]
   public_deps = [
     ":webrtc_opus_c",
     "../../rtc_base:protobuf_utils",
@@ skipping 74 matching lines @@
     "audio_network_adaptor/fec_controller_rplr_based.cc",
     "audio_network_adaptor/fec_controller_rplr_based.h",
     "audio_network_adaptor/frame_length_controller.cc",
     "audio_network_adaptor/frame_length_controller.h",
     "audio_network_adaptor/include/audio_network_adaptor.h",
     "audio_network_adaptor/util/threshold_curve.h",
   ]
 
   deps = [
     "../..:webrtc_common",
+    "../../api:optional",
     "../../common_audio",
     "../../logging:rtc_event_log_api",
     "../../rtc_base:protobuf_utils",
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers",
   ]
 
   if (rtc_enable_protobuf) {
     deps += [
       ":ana_config_proto",
       ":ana_debug_dump_proto",
     ]
   }
 
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 }
 
 rtc_source_set("neteq_decoder_enum") {
   sources = [
     "neteq/neteq_decoder_enum.cc",
     "neteq/neteq_decoder_enum.h",
   ]
   deps = [
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../rtc_base:rtc_base_approved",
   ]
 }
 
 rtc_static_library("neteq") {
   sources = [
     "neteq/accelerate.cc",
     "neteq/accelerate.h",
     "neteq/audio_decoder_impl.cc",
@@ skipping 69 matching lines @@
 
   deps = [
     ":audio_coding_module_typedefs",
     ":cng",
     ":g711",
     ":isac_fix",
     ":neteq_decoder_enum",
     ":pcm16b",
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../rtc_base:gtest_prod",
     "../../rtc_base:rtc_base_approved",
     "../../system_wrappers",
   ]
 
   defines = []
 
   if (rtc_include_ilbc) {
@@ skipping 39 matching lines @@
 
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     ":neteq",
     "..:module_api",
     "../..:webrtc_common",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../api/audio_codecs:builtin_audio_decoder_factory",
     "../../rtc_base:rtc_base_approved",
     "../rtp_rtcp",
   ]
 }
 
 rtc_source_set("neteq_test_tools") {
   testonly = true
   sources = [
@@ skipping 64 matching lines @@
   public_configs = [ ":neteq_tools_config" ]
 
   if (!build_with_chromium && is_clang) {
     # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     "../..:webrtc_common",
     "../../api:array_view",
+    "../../api:optional",
     "../../api/audio_codecs:audio_codecs_api",
     "../../common_audio",
     "../../rtc_base:rtc_base_approved",
     "../rtp_rtcp",
   ]
 
   public_deps = [
     ":neteq_tools_minimal",
   ]
 }
@@ skipping 104 matching lines @@
       "test/utility.cc",
       "test/utility.h",
     ]
     deps = [
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":audio_format_conversion",
       ":pcm16b_c",
       "..:module_api",
       "../..:webrtc_common",
+      "../../api:optional",
       "../../api/audio_codecs:builtin_audio_decoder_factory",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers",
       "../../test:test_support",
     ]
     defines = audio_coding_defines
     if (is_win) {
       cflags = [
         # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
         "/wd4373",  # virtual function override.
@@ skipping 86 matching lines @@
       "test/utility.cc",
       "test/utility.h",
     ]
 
     deps = [
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../api:optional",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
     ]
   }  # delay_test
 
   rtc_executable("insert_packet_with_timing") {
     testonly = true
     sources = [
       "test/Channel.cc",
       "test/Channel.h",
       "test/PCMFile.cc",
       "test/PCMFile.h",
       "test/insert_packet_with_timing.cc",
     ]
 
     if (!build_with_chromium && is_clang) {
       # Suppress warnings from the Chromium Clang plugin (bugs.webrtc.org/163).
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
 
     deps = [
       ":audio_coding",
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../api:optional",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
     ]
   }  # insert_packet_with_timing
 
   audio_decoder_unittests_resources =
@@ skipping 805 matching lines @@
 # webrtc/api/audio_codecs:builtin_audio_decoder_factory instead.
 # TODO(ossu): Remove this.
 rtc_source_set("builtin_audio_encoder_factory") {
   sources = [
     "codecs/builtin_audio_encoder_factory.h",
   ]
   deps = [
     "../../api/audio_codecs:builtin_audio_encoder_factory",
   ]
 }