Step 1 to prepare call_test.* for combined audio/video tests.

webrtc/call/packet_injection_tests.cc

 /*
  *  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.
  */
 
@@ skipping 24 matching lines @@
 void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,
                                                 uint8_t payload_type,
                                                 const uint8_t* packet,
                                                 size_t length) {
   CreateSenderCall(Call::Config());
   CreateReceiverCall(Call::Config());
 
   test::NullTransport null_transport;
   CreateSendConfig(1, &null_transport);
   CreateMatchingReceiveConfigs(&null_transport);
-  receive_configs_[0].decoders[0].payload_type = payload_type;
+  video_receive_configs_[0].decoders[0].payload_type = payload_type;
   switch (codec_type) {
     case CodecType::kVp8:
-      receive_configs_[0].decoders[0].payload_name = "VP8";
+      video_receive_configs_[0].decoders[0].payload_name = "VP8";
       break;
     case CodecType::kH264:
-      receive_configs_[0].decoders[0].payload_name = "H264";
+      video_receive_configs_[0].decoders[0].payload_name = "H264";
       break;
   }
   CreateStreams();
 
   RTPHeader header;
   EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
   EXPECT_EQ(kSendSsrcs[0], header.ssrc)
       << "Packet should have configured SSRC to not be dropped early.";
   EXPECT_EQ(payload_type, header.payloadType);
   Start();
@@ skipping 20 matching lines @@
                              0xED,
                              0x58,
                              0xCB,
                              0xED,
                              0xDF};
 
   InjectIncorrectPacket(CodecType::kH264, 101, kPacket, sizeof(kPacket));
 }
 
 }  // namespace webrtc