Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/video/video_send_stream_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>  // max
@@ skipping 421 matching lines @@
   }
 
   test::PacketTransport* CreateSendTransport(Call* sender_call) override {
     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
     // Configure some network delay.
     const int kNetworkDelayMs = 100;
     FakeNetworkPipe::Config config;
     config.loss_percent = 5;
     config.queue_delay_ms = kNetworkDelayMs;
     return new test::PacketTransport(sender_call, this,
-                                     test::PacketTransport::kSender, config);
+                                     test::PacketTransport::kSender,
+                                     MediaType::VIDEO, config);
   }
 
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override {
     if (use_nack_) {
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms =
               VideoSendStreamTest::kNackRtpHistoryMs;
@@ skipping 134 matching lines @@
   }
 
   test::PacketTransport* CreateSendTransport(Call* sender_call) override {
     // At low RTT (< kLowRttNackMs) -> NACK only, no FEC.
     // Therefore we need some network delay.
     const int kNetworkDelayMs = 100;
     FakeNetworkPipe::Config config;
     config.loss_percent = 5;
     config.queue_delay_ms = kNetworkDelayMs;
     return new test::PacketTransport(sender_call, this,
-                                     test::PacketTransport::kSender, config);
+                                     test::PacketTransport::kSender,
+                                     MediaType::VIDEO, config);
   }
 
   void ModifyVideoConfigs(
       VideoSendStream::Config* send_config,
       std::vector<VideoReceiveStream::Config>* receive_configs,
       VideoEncoderConfig* encoder_config) override {
     if (use_nack_) {
       send_config->rtp.nack.rtp_history_ms =
           (*receive_configs)[0].rtp.nack.rtp_history_ms =
               VideoSendStreamTest::kNackRtpHistoryMs;
@@ skipping 648 matching lines @@
       return SEND_PACKET;
     }
 
     test::PacketTransport* CreateSendTransport(Call* sender_call) override {
       const int kNetworkDelayMs = 50;
       FakeNetworkPipe::Config config;
       config.loss_percent = 10;
       config.link_capacity_kbps = kCapacityKbps;
       config.queue_delay_ms = kNetworkDelayMs;
       return new test::PacketTransport(sender_call, this,
-                                       test::PacketTransport::kSender, config);
+                                       test::PacketTransport::kSender,
+                                       MediaType::VIDEO, config);
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         VideoEncoderConfig* encoder_config) override {
       // Turn on RTX.
       send_config->rtp.rtx.payload_type = kFakeVideoSendPayloadType;
       send_config->rtp.rtx.ssrcs.push_back(kVideoSendSsrcs[0]);
     }
@@ skipping 2007 matching lines @@
     rtc::CriticalSection crit_;
     uint32_t max_bitrate_bps_ GUARDED_BY(&crit_);
     bool first_packet_sent_ GUARDED_BY(&crit_);
     rtc::Event bitrate_changed_event_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc