Split audio mixer into interface and implementation.

webrtc/modules/BUILD.gn

 # Copyright (c) 2016 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("../build/webrtc.gni")
 import("audio_coding/audio_coding.gni")
 
 declare_args() {
   # Desktop capturer is supported only on Windows, OSX and Linux.
   rtc_desktop_capture_supported = is_win || is_mac || is_linux
 }
 
 group("modules") {
   public_deps = [
     "audio_coding",
     "audio_conference_mixer",
     "audio_device",
-    "audio_mixer",
+    "audio_mixer:audio_mixer_impl",
     "audio_processing",
     "bitrate_controller",
     "desktop_capture",
     "media_file",
     "rtp_rtcp",
     "utility",
     "video_coding",
     "video_processing",
   ]
 }
@@ skipping 588 matching lines @@
 
     deps += [
       ":audio_network_adaptor_unittests",
       "..:webrtc_common",
       "../base:rtc_base",  # TODO(kjellander): Cleanup in bugs.webrtc.org/3806.
       "../common_audio",
       "../common_video",
       "../system_wrappers:system_wrappers",
       "../test:rtp_test_utils",
       "../test:test_common",
+      "../test:test_support",

[aleloi, 2016/10/13 09:25:22]

This line reduced the output of "gn check <out_dir>
webrtc/modules:modules_unittests" from about 4000 lines to about 800.

       "../test:test_support_main",
       "../test:video_test_common",
       "audio_coding",
       "audio_coding:acm_receive_test",
       "audio_coding:acm_send_test",
       "audio_coding:builtin_audio_decoder_factory",
       "audio_coding:cng",
       "audio_coding:isac_fix",
       "audio_coding:neteq",
       "audio_coding:neteq_test_support",
       "audio_coding:neteq_unittest_tools",
       "audio_coding:pcm16b",
       "audio_coding:red",
       "audio_coding:webrtc_opus",
       "audio_conference_mixer",
       "audio_device",
-      "audio_mixer",
+      "audio_mixer:audio_frame_manipulator",
+      "audio_mixer:audio_mixer_impl",
       "audio_processing",
       "audio_processing:audioproc_test_utils",
       "bitrate_controller",
       "media_file",
       "pacing",
       "remote_bitrate_estimator",
       "remote_bitrate_estimator:bwe_simulator_lib",
       "rtp_rtcp",
       "utility",
       "video_capture",
@@ skipping 75 matching lines @@
       "../test:test_common",
       "../test:test_support_main",
       "remote_bitrate_estimator:bwe_simulator_lib",
       "video_capture",
       "//testing/gmock",
       "//testing/gtest",
       "//third_party/gflags",
     ]
   }
 }