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

webrtc/video/video_quality_test.cc

Index: webrtc/video/video_quality_test.cc
diff --git a/webrtc/video/video_quality_test.cc b/webrtc/video/video_quality_test.cc
index d09d2a2bf9de77eda7afc8e8ea093e9fb92bb3b5..2867da7ce085ea70ec156dbd61f2d40d724d0b6e 100644
--- a/webrtc/video/video_quality_test.cc
+++ b/webrtc/video/video_quality_test.cc
@@ -781,7 +781,7 @@ void VideoQualityTest::SetupCommon(Transport* send_transport,
     trace_to_stderr_.reset(new test::TraceToStderr);
 
   size_t num_streams = params_.ss.streams.size();
-  CreateSendConfig(num_streams, send_transport);
+  CreateSendConfig(num_streams, 0, send_transport);
 
   int payload_type;
   if (params_.common.codec == "VP8") {
@@ -817,7 +817,7 @@ void VideoQualityTest::SetupCommon(Transport* send_transport,
   video_encoder_config_.streams = params_.ss.streams;
   video_encoder_config_.spatial_layers = params_.ss.spatial_layers;
 
-  CreateMatchingReceiveConfigs(recv_transport);
+  CreateMatchingReceiveConfigs(false, recv_transport);
 
   for (size_t i = 0; i < num_streams; ++i) {
     video_receive_configs_[i].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
@@ -964,7 +964,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
       disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
       params_.analyzer.test_durations_secs * params_.common.fps,
       graph_data_output_file, graph_title,
-      kSendSsrcs[params_.ss.selected_stream]);
+      kVideoSendSsrcs[params_.ss.selected_stream]);
 
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
@@ -979,7 +979,7 @@ void VideoQualityTest::RunWithAnalyzer(const Params& params) {
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
-  CreateStreams();
+  CreateVideoStreams();
   analyzer.input_ = video_send_stream_->Input();
   analyzer.send_stream_ = video_send_stream_;