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

webrtc/test/fake_network_pipe.h

 /*
  *  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.
  */
 
 #ifndef WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
 #define WEBRTC_TEST_FAKE_NETWORK_PIPE_H_
 
 #include <memory>
 #include <set>
 #include <string.h>
 #include <queue>
 
 #include "webrtc/base/constructormagic.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/random.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class Clock;
 class PacketReceiver;
+enum class MediaType;
 
 class NetworkPacket {
  public:
   NetworkPacket(const uint8_t* data,
                 size_t length,
                 int64_t send_time,
                 int64_t arrival_time)
       : data_(new uint8_t[length]),
         data_length_(length),
         send_time_(send_time),
@@ skipping 37 matching lines @@
     // Link capacity in kbps.
     int link_capacity_kbps = 0;
     // Random packet loss.
     int loss_percent = 0;
     // If packets are allowed to be reordered.
     bool allow_reordering = false;
     // The average length of a burst of lost packets.
     int avg_burst_loss_length = -1;
   };
 
-  FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config);
+  FakeNetworkPipe(Clock* clock, const FakeNetworkPipe::Config& config,
+                  MediaType media_type);

[perkj_webrtc, 2017/03/30 09:13:24]

This will not work for tests that mux audio and video on the same transport. The
fake network pipe then needs to know per packet what type it is. 

[nisse-webrtc, 2017/03/30 09:22:35]

For webrtc tests, I fixed that case by adding a payload-type demuxer, see this
cl's changes to call_test.cc.

Letting the network pipe keep track of media type per packet would make sense to
me, but I haven't looked into how to make that work. We'd then need media type
config at all places where packets are inserted into the pipe.

   FakeNetworkPipe(Clock* clock,
-                  const FakeNetworkPipe::Config& config,
+                  const FakeNetworkPipe::Config& config, MediaType media_type,
                   uint64_t seed);
   ~FakeNetworkPipe();
 
   // Must not be called in parallel with SendPacket or Process.
   void SetReceiver(PacketReceiver* receiver);
 
   // Sets a new configuration. This won't affect packets already in the pipe.
   void SetConfig(const FakeNetworkPipe::Config& config);
 
   // Sends a new packet to the link.
   void SendPacket(const uint8_t* packet, size_t packet_length);
 
   // Processes the network queues and trigger PacketReceiver::IncomingPacket for
   // packets ready to be delivered.
   void Process();
   int64_t TimeUntilNextProcess() const;
 
   // Get statistics.
   float PercentageLoss();
   int AverageDelay();
   size_t dropped_packets() { return dropped_packets_; }
   size_t sent_packets() { return sent_packets_; }
 
  private:
   Clock* const clock_;
+  const MediaType media_type_;
   rtc::CriticalSection lock_;
   PacketReceiver* packet_receiver_;
   std::queue<NetworkPacket*> capacity_link_;
   Random random_;
 
   // Since we need to access both the packet with the earliest and latest
   // arrival time we need to use a multiset to keep all packets sorted,
   // hence, we cannot use a priority queue.
   struct PacketArrivalTimeComparator {
     bool operator()(const NetworkPacket* p1, const NetworkPacket* p2) {
@@ skipping 23 matching lines @@
   int64_t next_process_time_;
 
   int64_t last_log_time_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(FakeNetworkPipe);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_FAKE_NETWORK_PIPE_H_