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..79f1ff69473846f09d355b2a5e85b62377feae98 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -284,7 +284,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
 
   test::FakeDecoder fake_decoder;
 
-  CreateSendConfig(1, &sync_send_transport);
+  CreateSendConfig(1, 0, &sync_send_transport);
   CreateMatchingReceiveConfigs(&sync_receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
@@ -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);
   }