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

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.
  */
 
@@ skipping 175 matching lines @@
     const MediaType media_type_;
 
     RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaTypePacketReceiver);
   };
 
   FakeNetworkPipe::Config audio_net_config;
   audio_net_config.queue_delay_ms = 500;
   audio_net_config.loss_percent = 5;
   test::PacketTransport audio_send_transport(sender_call_.get(), &observer,
                                              test::PacketTransport::kSender,
+                                             MediaType::AUDIO,
                                              audio_net_config);
   MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(),
                                          MediaType::AUDIO);
   audio_send_transport.SetReceiver(&audio_receiver);
 
   test::PacketTransport video_send_transport(sender_call_.get(), &observer,
                                              test::PacketTransport::kSender,
+                                             MediaType::VIDEO,
                                              FakeNetworkPipe::Config());
   MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(),
                                          MediaType::VIDEO);
   video_send_transport.SetReceiver(&video_receiver);
 
   test::PacketTransport receive_transport(
       receiver_call_.get(), &observer, test::PacketTransport::kReceiver,
+      MediaType::VIDEO,
       FakeNetworkPipe::Config());
   receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
   CreateSendConfig(1, 0, 0, &video_send_transport);
   CreateMatchingReceiveConfigs(&receive_transport);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
@@ skipping 116 matching lines @@
           start_time_ms_(start_time_ms),
           run_time_ms_(run_time_ms),
           creation_time_ms_(clock_->TimeInMilliseconds()),
           capturer_(nullptr),
           rtp_start_timestamp_set_(false),
           rtp_start_timestamp_(0) {}
 
    private:
     test::PacketTransport* CreateSendTransport(Call* sender_call) override {
       return new test::PacketTransport(
-          sender_call, this, test::PacketTransport::kSender, net_config_);
+          sender_call, this, test::PacketTransport::kSender, MediaType::VIDEO,
+          net_config_);
     }
 
     test::PacketTransport* CreateReceiveTransport() override {
       return new test::PacketTransport(
-          nullptr, this, test::PacketTransport::kReceiver, net_config_);
+          nullptr, this, test::PacketTransport::kReceiver, MediaType::VIDEO,
+          net_config_);
     }
 
     void OnFrame(const VideoFrame& video_frame) override {
       rtc::CritScope lock(&crit_);
       if (video_frame.ntp_time_ms() <= 0) {
         // Haven't got enough RTCP SR in order to calculate the capture ntp
         // time.
         return;
       }
 
@@ skipping 377 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc