Made AudioReceiveStream a mixer participant.

webrtc/audio/BUILD.gn

 # Copyright (c) 2015 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")
 
@@ skipping 11 matching lines @@
 
   if (is_clang) {
     # Suppress warnings from Chrome's Clang plugins.
     # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
     suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
   }
 
   deps = [
     "..:webrtc_common",
     "../api:call_api",
+    "../modules/audio_mixer",

[the sun, 2016/10/03 11:41:26]

Should really just depend on interface here - not the whole module, but that
requires changing the structure first.

     "../system_wrappers",
     "../voice_engine",
   ]
 }
 if (rtc_include_tests) {
   rtc_source_set("audio_tests") {
     testonly = true
     sources = [
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
       "audio_state_unittest.cc",
     ]
     deps = [
       ":audio",
       "//testing/gmock",
       "//testing/gtest",
     ]
     if (is_clang) {
       # Suppress warnings from the Chromium Clang plugin.
       # See http://code.google.com/p/webrtc/issues/detail?id=163 for details.
       suppressed_configs += [ "//build/config/clang:find_bad_constructs" ]
     }
   }
 }