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

webrtc/call/call_perf_tests.cc

Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index faefc425d7c7bc6bd4e7149757e5b2006f21731f..0ec96c1c0b69543f0f456860cb425644bbaaac29 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -284,8 +284,8 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1, &sync_send_transport);
-  CreateMatchingReceiveConfigs(&sync_receive_transport);
+  CreateSendConfig(1, 0, &sync_send_transport);
+  CreateMatchingReceiveConfigs(false, &sync_receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
@@ -318,9 +318,9 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
   if (create_audio_first) {
     audio_receive_stream =
         receiver_call_->CreateAudioReceiveStream(audio_recv_config);
-    CreateStreams();
+    CreateVideoStreams();
   } else {
-    CreateStreams();
+    CreateVideoStreams();
     audio_receive_stream =
         receiver_call_->CreateAudioReceiveStream(audio_recv_config);
   }