MockAecDump and integration tests between AecDump and AudioProcessing

webrtc/modules/audio_processing/aec_dump/BUILD.gn

Index: webrtc/modules/audio_processing/aec_dump/BUILD.gn
diff --git a/webrtc/modules/audio_processing/aec_dump/BUILD.gn b/webrtc/modules/audio_processing/aec_dump/BUILD.gn
index 4a02299eaa0d2ce2143b805ec7901726448ff7d3..950dd68b176a49485d213e032c6c06db12ff6725 100644
--- a/webrtc/modules/audio_processing/aec_dump/BUILD.gn
+++ b/webrtc/modules/audio_processing/aec_dump/BUILD.gn
@@ -1,3 +1,11 @@
+# Copyright (c) 2017 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")  # This contains def of 'rtc_enable_protobuf'
 
 rtc_source_set("aec_dump") {
@@ -14,6 +22,38 @@ rtc_source_set("aec_dump") {
   ]
 }
 
+rtc_source_set("mock_aec_dump") {
+  testonly = true
+  sources = [
+    "mock_aec_dump.cc",
+    "mock_aec_dump.h",
+  ]
+
+  deps = [
+    "..:aec_dump_interface",
+  ]
+  public_deps = [
+    "../..:module_api",
+    "../../../test:test_support",
+    "//testing/gmock",
+  ]
+}
+
+rtc_source_set("mock_aec_dump_unittests") {
+  testonly = true
+
+  sources = [
+    "aec_dump_integration_test.cc",
+  ]
+
+  deps = [
+    ":mock_aec_dump",
+    "..:audio_processing",
+    "../../../base:rtc_base_approved",
+    "//testing/gtest",
+  ]
+}
+
 if (rtc_enable_protobuf) {
   rtc_source_set("aec_dump_impl") {
     sources = [