Store/restore RTP state for audio streams with same SSRC within a call

webrtc/call/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")
 
@@ skipping 75 matching lines @@
     }
     sources = [
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
       "rtx_receive_stream_unittest.cc",
     ]
     deps = [
       ":call",
+      "../api:mock_audio_mixer",
       "../base:rtc_base_approved",
       "../logging:rtc_event_log_api",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../modules/bitrate_controller",
       "../modules/congestion_controller:mock_congestion_controller",
       "../modules/pacing",
       "../modules/rtp_rtcp",
       "../modules/rtp_rtcp:mock_rtp_rtcp",
       "../system_wrappers",
@@ skipping 45 matching lines @@
       "//testing/gtest",
       "//webrtc/test:field_trial",
       "//webrtc/test:test_common",
     ]
     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" ]
     }
   }
 }