Reland of Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/call/rampup_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.
  */
 
@@ skipping 71 matching lines @@
   call_config.bitrate_config.min_bitrate_bps = 10000;
   return call_config;
 }
 
 void RampUpTester::OnVideoStreamsCreated(
     VideoSendStream* send_stream,
     const std::vector<VideoReceiveStream*>& receive_streams) {
   send_stream_ = send_stream;
 }
 
+MediaType RampUpTester::SelectMediaType() {
+  if (num_video_streams_ > 0) {
+    if (num_audio_streams_ > 0) {
+      // Rely on call to set media type from payload type.
+      return MediaType::ANY;
+    } else {
+      return MediaType::VIDEO;
+    }
+  } else {
+    return MediaType::AUDIO;
+  }
+}
+
 test::PacketTransport* RampUpTester::CreateSendTransport(Call* sender_call) {
   send_transport_ = new test::PacketTransport(sender_call, this,
                                               test::PacketTransport::kSender,
+                                              SelectMediaType(),
                                               forward_transport_config_);
   return send_transport_;
 }
 
+test::PacketTransport* RampUpTester::CreateReceiveTransport() {
+  return new test::PacketTransport(nullptr, this,
+                                   test::PacketTransport::kReceiver,
+                                   SelectMediaType(),
+                                   FakeNetworkPipe::Config());
+}
+
 size_t RampUpTester::GetNumVideoStreams() const {
   return num_video_streams_;
 }
 
 size_t RampUpTester::GetNumAudioStreams() const {
   return num_audio_streams_;
 }
 
 size_t RampUpTester::GetNumFlexfecStreams() const {
   return num_flexfec_streams_;
@@ skipping 534 matching lines @@
   RunBaseTest(&test);
 }
 
 TEST_F(RampUpTest, AudioTransportSequenceNumber) {
   RampUpTester test(0, 1, 0, 300000, 10000,
                     RtpExtension::kTransportSequenceNumberUri, false, false,
                     false);
   RunBaseTest(&test);
 }
 }  // namespace webrtc