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

webrtc/call/packet_injection_tests.cc

Index: webrtc/call/packet_injection_tests.cc
diff --git a/webrtc/call/packet_injection_tests.cc b/webrtc/call/packet_injection_tests.cc
index 315fc7b8f24d808ef8b4c4cf05740cd3d55dda00..5cba1d9ababde45e81701f30218085b2bb655b08 100644
--- a/webrtc/call/packet_injection_tests.cc
+++ b/webrtc/call/packet_injection_tests.cc
@@ -40,8 +40,8 @@ void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,
   CreateReceiverCall(Call::Config());
 
   test::NullTransport null_transport;
-  CreateSendConfig(1, &null_transport);
-  CreateMatchingReceiveConfigs(&null_transport);
+  CreateSendConfig(1, 0, &null_transport);
+  CreateMatchingReceiveConfigs(false, &null_transport);
   video_receive_configs_[0].decoders[0].payload_type = payload_type;
   switch (codec_type) {
     case CodecType::kVp8:
@@ -51,11 +51,11 @@ void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,
       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();