Conversational Speech tool, MultiEndCall class and unit tests via mocking

webrtc/modules/audio_processing/test/conversational_speech/BUILD.gn

 # 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/webrtc.gni")
 
@@ skipping 15 matching lines @@
     "//webrtc/base:rtc_base_approved",
     "//webrtc/test:test_support",
   ]
 }
 
 rtc_static_library("lib") {
   testonly = true
   sources = [
     "config.cc",
     "config.h",
+    "multiend_call.cc",
+    "multiend_call.h",
     "timing.cc",
     "timing.h",
+    "wavreader_abstract_factory.h",
+    "wavreader_adaptor.cc",
+    "wavreader_adaptor.h",
+    "wavreader_factory.cc",
+    "wavreader_factory.h",
+    "wavreader_interface.h",
   ]
   deps = [
+    "//webrtc:webrtc_common",
     "//webrtc/base:rtc_base_approved",
+    "//webrtc/common_audio",
   ]
   visibility = [ ":*" ]  # Only targets in this file can depend on this.
 }
 
 rtc_source_set("unittest") {
   testonly = true
   sources = [
     "generator_unittest.cc",
+    "mock_wavreader.h",
+    "mock_wavreader_factory.cc",
+    "mock_wavreader_factory.h",
   ]
   deps = [
     ":lib",
     "//testing/gmock",
     "//testing/gtest",
+    "//webrtc:webrtc_common",
     "//webrtc/test:test_support",
   ]
 }