Add audio streams to CallTest and a first A/V call test.

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 22 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);
+  CreateSendConfig(1, 0, &null_transport);
   CreateMatchingReceiveConfigs(&null_transport);
   video_receive_configs_[0].decoders[0].payload_type = payload_type;
   switch (codec_type) {
     case CodecType::kVp8:
       video_receive_configs_[0].decoders[0].payload_name = "VP8";
       break;
     case CodecType::kH264:
       video_receive_configs_[0].decoders[0].payload_name = "H264";
       break;
   }
-  CreateStreams();
+  CreateVideoStreams();
 
   RTPHeader header;
   EXPECT_TRUE(rtp_header_parser_->Parse(packet, length, &header));
-  EXPECT_EQ(kSendSsrcs[0], header.ssrc)
+  EXPECT_EQ(kVideoSendSsrcs[0], header.ssrc)
       << "Packet should have configured SSRC to not be dropped early.";
   EXPECT_EQ(payload_type, header.payloadType);
   Start();
   EXPECT_EQ(PacketReceiver::DELIVERY_PACKET_ERROR,
             receiver_call_->Receiver()->DeliverPacket(MediaType::VIDEO, packet,
                                                       length, PacketTime()));
   Stop();
 
   DestroyStreams();
 }
@@ skipping 13 matching lines @@
                              0xED,
                              0x58,
                              0xCB,
                              0xED,
                              0xDF};
 
   InjectIncorrectPacket(CodecType::kH264, 101, kPacket, sizeof(kPacket));
 }
 
 }  // namespace webrtc