Don't hardcode MediaType::ANY in FakeNetworkPipe.

webrtc/test/fake_network_pipe.cc

 /*
  *  Copyright (c) 2012 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 10 matching lines @@
 #include "webrtc/call/call.h"
 #include "webrtc/system_wrappers/include/clock.h"
 
 namespace webrtc {
 
 namespace {
 constexpr int64_t kDefaultProcessIntervalMs = 5;
 }
 
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
-                                 const FakeNetworkPipe::Config& config)
-    : FakeNetworkPipe(clock, config, 1) {}
+                                 const FakeNetworkPipe::Config& config,
+                                 MediaType media_type)
+    : FakeNetworkPipe(clock, config, media_type, 1) {}
 
 FakeNetworkPipe::FakeNetworkPipe(Clock* clock,
                                  const FakeNetworkPipe::Config& config,
+                                 MediaType media_type,
                                  uint64_t seed)
     : clock_(clock),
+      media_type_(media_type),
       packet_receiver_(NULL),
       random_(seed),
       config_(),
       dropped_packets_(0),
       sent_packets_(0),
       total_packet_delay_(0),
       bursting_(false),
       next_process_time_(clock_->TimeInMilliseconds()),
       last_log_time_(clock_->TimeInMilliseconds()) {
   SetConfig(config);
@@ skipping 144 matching lines @@
       // |time_now| might be later than when the packet should have arrived, due
       // to NetworkProcess being called too late. For stats, use the time it
       // should have been on the link.
       total_packet_delay_ += packet->arrival_time() - packet->send_time();
     }
     sent_packets_ += packets_to_deliver.size();
   }
   while (!packets_to_deliver.empty()) {
     NetworkPacket* packet = packets_to_deliver.front();
     packets_to_deliver.pop();
-    packet_receiver_->DeliverPacket(MediaType::ANY, packet->data(),
+    packet_receiver_->DeliverPacket(media_type_, packet->data(),
                                     packet->data_length(), PacketTime());
     delete packet;
   }
 
   next_process_time_ = !delay_link_.empty()
                            ? (*delay_link_.begin())->arrival_time()
                            : time_now + kDefaultProcessIntervalMs;
 }
 
 int64_t FakeNetworkPipe::TimeUntilNextProcess() const {
   rtc::CritScope crit(&lock_);
   return std::max<int64_t>(next_process_time_ - clock_->TimeInMilliseconds(),
                            0);
 }
 
 }  // namespace webrtc