Move RtpExtension to api/ directory and config.h/.cc to call/.

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 1371 matching lines @@
       "test/utility.cc",
       "test/utility.h",
     ]
 
     deps = [
       ":audio_coding",
       ":audio_coding_module_typedefs",
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
   }  # delay_test
 
@@ skipping 10 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 = [
       ":audio_coding",
       ":audio_format_conversion",
       "..:module_api",
       "../../:webrtc_common",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "../rtp_rtcp",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
   }  # insert_packet_with_timing
 
@@ skipping 27 matching lines @@
 
     deps += audio_coding_deps
     deps += [
       ":ilbc",
       ":isac",
       ":isac_fix",
       ":neteq",
       ":neteq_tools",
       "../../api/audio_codecs:audio_codecs_api",
       "../../api/audio_codecs/opus:audio_encoder_opus",
+      "../../call:call",
       "../../common_audio",
       "../../rtc_base:protobuf_utils",
       "../../test:test_main",
       "//testing/gtest",
     ]
 
     data = audio_decoder_unittests_resources
 
     if (is_android) {
       deps += [ "//testing/android/native_test:native_test_native_code" ]
@@ skipping 31 matching lines @@
         cflags = [
           # TODO(kjellander): bugs.webrtc.org/261: Fix this warning.
           "/wd4373",  # virtual function override.
         ]
       }
 
       deps += [
         ":neteq",
         ":neteq_test_tools",
         "../..:webrtc_common",
+        "../../call:call",
         "../../rtc_base:rtc_base_approved",
         "../../system_wrappers:system_wrappers_default",
         "../../test:test_support",
         "//third_party/gflags",
       ]
     }
   }
 
   audio_codec_speed_tests_resources = [
     "//resources/audio_coding/music_stereo_48kHz.pcm",
@@ skipping 35 matching lines @@
     }
 
     if (is_ios) {
       deps += [ ":audio_codec_speed_tests_data" ]
     }
 
     deps += [
       ":isac_fix",
       ":webrtc_opus",
       "../..:webrtc_common",
+      "../../api:libjingle_peerconnection_api",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
+      "../../system_wrappers:metrics_default",
       "../../system_wrappers:system_wrappers_default",
+      "../../test:field_trial",
       "../../test:test_main",
       "../audio_processing",
       "//testing/gtest",
     ]
   }
 
   rtc_source_set("neteq_test_support") {
     testonly = true
     sources = [
       "neteq/tools/neteq_external_decoder_test.cc",
@@ skipping 110 matching lines @@
       ":cng",
       ":g711",
       ":g722",
       ":ilbc",
       ":isac",
       ":neteq",
       ":neteq_test_tools_deprecated",
       ":pcm16b",
       ":webrtc_opus",
       "../..:webrtc_common",
+      "../../api:libjingle_peerconnection_api",
+      "../../call:call",
       "../../common_audio",
       "../../rtc_base:rtc_base_approved",
+      "../../system_wrappers:metrics_default",
+      "../../test:field_trial",
     ]
 
     configs += [ ":RTPencode_config" ]
 
     sources = [
       "neteq/test/PayloadTypes.h",
       "neteq/test/RTPencode.cc",
     ]
 
     include_dirs = [
@@ skipping 22 matching lines @@
   }
 
   rtc_executable("rtpcat") {
     testonly = true
 
     sources = [
       "neteq/tools/rtpcat.cc",
     ]
 
     deps = [
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:rtp_test_utils",
       "//testing/gtest",
     ]
   }
 
   rtc_executable("RTPtimeshift") {
     testonly = true
 
@@ skipping 25 matching lines @@
     testonly = true
 
     sources = [
       "neteq/tools/rtp_analyze.cc",
     ]
 
     deps = [
       ":neteq",
       ":neteq_test_tools",
       ":pcm16b",
+      "../../call:call",
       "../../system_wrappers:system_wrappers_default",
       "//testing/gtest",
       "//third_party/gflags:gflags",
     ]
 
     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_executable("neteq_opus_quality_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_opus_quality_test.cc",
     ]
 
     deps = [
       ":neteq",
       ":neteq_quality_test_support",
       ":neteq_tools",
       ":webrtc_opus",
+      "../../call:call",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("neteq_speed_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_speed_test.cc",
     ]
 
     deps = [
       ":neteq",
       ":neteq_test_support",
       "../..:webrtc_common",
+      "../../call:call",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_support",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("neteq_ilbc_quality_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_ilbc_quality_test.cc",
     ]
 
     deps = [
       ":ilbc",
       ":neteq",
       ":neteq_quality_test_support",
       ":neteq_tools",
       "../..:webrtc_common",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../system_wrappers:system_wrappers_default",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("neteq_isac_quality_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_isac_quality_test.cc",
     ]
 
     deps = [
       ":isac_fix",
       ":neteq",
       ":neteq_quality_test_support",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("neteq_pcmu_quality_test") {
     testonly = true
 
     sources = [
       "neteq/test/neteq_pcmu_quality_test.cc",
     ]
 
     deps = [
       ":g711",
       ":neteq",
       ":neteq_quality_test_support",
+      "../../call:call",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 
   rtc_executable("isac_fix_test") {
     testonly = true
 
@@ skipping 132 matching lines @@
 
   rtc_executable("webrtc_opus_fec_test") {
     testonly = true
 
     sources = [
       "codecs/opus/opus_fec_test.cc",
     ]
 
     deps = [
       ":webrtc_opus",
+      "../../call:call",
       "../../common_audio",
       "../../rtc_base:rtc_base_approved",
       "../../test:test_main",
       "//testing/gtest",
     ]
 
     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" ]
     }
@@ skipping 185 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",
   ]
 }