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

webrtc/call/bitrate_estimator_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 <functional>
@@ skipping 102 matching lines @@
     config.audio_state = AudioState::Create(audio_state_config);
     receiver_call_.reset(Call::Create(config));
     sender_call_.reset(Call::Create(config));
 
     send_transport_.reset(new test::DirectTransport(sender_call_.get()));
     send_transport_->SetReceiver(receiver_call_->Receiver());
     receive_transport_.reset(new test::DirectTransport(receiver_call_.get()));
     receive_transport_->SetReceiver(sender_call_->Receiver());
 
     video_send_config_ = VideoSendStream::Config(send_transport_.get());
-    video_send_config_.rtp.ssrcs.push_back(kSendSsrcs[0]);
+    video_send_config_.rtp.ssrcs.push_back(kVideoSendSsrcs[0]);
     // Encoders will be set separately per stream.
     video_send_config_.encoder_settings.encoder = nullptr;
     video_send_config_.encoder_settings.payload_name = "FAKE";
-    video_send_config_.encoder_settings.payload_type = kFakeSendPayloadType;
+    video_send_config_.encoder_settings.payload_type =
+        kFakeVideoSendPayloadType;
     video_encoder_config_.streams = test::CreateVideoStreams(1);
 
     receive_config_ = VideoReceiveStream::Config(receive_transport_.get());
     // receive_config_.decoders will be set by every stream separately.
     receive_config_.rtp.remote_ssrc = video_send_config_.rtp.ssrcs[0];
-    receive_config_.rtp.local_ssrc = kReceiverLocalSsrc;
+    receive_config_.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     receive_config_.rtp.remb = true;
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kTOffset, kTOFExtensionId));
     receive_config_.rtp.extensions.push_back(
         RtpExtension(RtpExtension::kAbsSendTime, kASTExtensionId));
   }
 
   virtual void TearDown() {
     std::for_each(streams_.begin(), streams_.end(),
                   std::mem_fun(&Stream::StopSending));
@@ skipping 199 matching lines @@
       RtpExtension(RtpExtension::kTOffset, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   receiver_log_.PushExpectedLogLine(kSingleStreamLog);
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc