Replace RtpStreamReceiver::DeliverRtp with OnRtpPacket.

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 43 matching lines @@
 }
 
 if (rtc_include_tests) {
   rtc_source_set("call_tests") {
     testonly = true
     sources = [
       "bitrate_allocator_unittest.cc",
       "bitrate_estimator_tests.cc",
       "call_unittest.cc",
       "flexfec_receive_stream_unittest.cc",
-      "packet_injection_tests.cc",
     ]
     deps = [
       ":call",
       "../base:rtc_base_approved",
       "../modules/audio_device:mock_audio_device",
       "../modules/audio_mixer",
       "../test:test_common",
       "//testing/gmock",
       "//testing/gtest",
     ]
@@ skipping 13 matching lines @@
     deps = [
       "//testing/gtest",
       "//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" ]
     }
   }
 }