Introduce RtpStreamReceiver and RtpStreamReceiverControllerInterface.

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("../webrtc.gni")
 if (is_android) {
@@ skipping 61 matching lines @@
     sources = [
       "audio_receive_stream_unittest.cc",
       "audio_send_stream_unittest.cc",
       "audio_state_unittest.cc",
     ]
     deps = [
       ":audio",
       "../api:mock_audio_mixer",
       "../base:rtc_base_approved",
       "../base:rtc_task_queue",
+      "../call:rtp_receiver",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer:audio_mixer_impl",
       "../modules/congestion_controller:congestion_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing:pacing",
       "../test:test_common",
       "../test:test_support",
       "utility:utility_tests",
       "//testing/gmock",
       "//testing/gtest",
@@ skipping 31 matching lines @@
         "//resources/voice_engine/audio_tiny48.wav",
       ]
 
       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" ]
       }
     }
   }
 }