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

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
@@ skipping 266 matching lines @@
                                             test::PacketTransport::kSender,
                                             FakeNetworkPipe::Config());
   sync_send_transport.SetReceiver(receiver_call_->Receiver());
   test::PacketTransport sync_receive_transport(receiver_call_.get(), &observer,
                                                test::PacketTransport::kReceiver,
                                                FakeNetworkPipe::Config());
   sync_receive_transport.SetReceiver(sender_call_->Receiver());
 
   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);
   audio_send_config.voe_channel_id = send_channel_id;
   audio_send_config.rtp.ssrc = kAudioSendSsrc;
   AudioSendStream* audio_send_stream =
       sender_call_->CreateAudioSendStream(audio_send_config);
 
   CodecInst isac = {103, "ISAC", 16000, 480, 1, 32000};
   EXPECT_EQ(0, voe_codec->SetSendCodec(send_channel_id, isac));
@@ skipping 13 matching lines @@
   audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
   audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
   audio_recv_config.voe_channel_id = recv_channel_id;
   audio_recv_config.sync_group = kSyncGroup;
 
   AudioReceiveStream* audio_receive_stream;
 
   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);
   }
 
   CreateFrameGeneratorCapturer();
 
   Start();
 
   fake_audio_device.Start();
   EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
@@ skipping 401 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test, FakeNetworkPipe::Config());
 }
 
 }  // namespace webrtc