Support GCM ciphers even if ENABLE_EXTERNAL_AUTH is defined.

webrtc/p2p/base/fakepackettransport.h

 /*
  *  Copyright 2017 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_P2P_BASE_FAKEPACKETTRANSPORT_H_
 #define WEBRTC_P2P_BASE_FAKEPACKETTRANSPORT_H_
 
 #include <string>
 
 #include "webrtc/api/ortc/packettransportinterface.h"
 #include "webrtc/base/asyncinvoker.h"
 #include "webrtc/base/copyonwritebuffer.h"
+#include "webrtc/media/base/rtputils.h"

[joachim, 2017/02/28 00:11:45]

Same here.

 #include "webrtc/p2p/base/packettransportinternal.h"
 
 namespace rtc {
 
 // Used to simulate a packet-based transport.
 class FakePacketTransport : public PacketTransportInternal,
                             public webrtc::PacketTransportInterface {
  public:
   explicit FakePacketTransport(const std::string& debug_name)
       : debug_name_(debug_name) {}
@@ skipping 38 matching lines @@
                  size_t len,
                  const PacketOptions& options,
                  int flags) override {
     if (!dest_) {
       return -1;
     }
     CopyOnWriteBuffer packet(data, len);
     if (async_) {
       invoker_.AsyncInvokeDelayed<void>(
           RTC_FROM_HERE, Thread::Current(),
-          Bind(&FakePacketTransport::SendPacketInternal, this, packet),
+          Bind(&FakePacketTransport::SendPacketInternal, this, packet,
+              options),
           async_delay_ms_);
     } else {
-      SendPacketInternal(packet);
+      SendPacketInternal(packet, options);
     }
     SentPacket sent_packet(options.packet_id, TimeMillis());
     SignalSentPacket(this, sent_packet);
     return static_cast<int>(len);
   }
   int SetOption(Socket::Option opt, int value) override { return true; }
   bool GetOption(Socket::Option opt, int* value) override { return true; }
   int GetError() override { return 0; }
 
  protected:
@@ skipping 12 matching lines @@
   }
 
   void set_receiving(bool receiving) {
     if (receiving_ == receiving) {
       return;
     }
     receiving_ = receiving;
     SignalReceivingState(this);
   }
 
-  void SendPacketInternal(const CopyOnWriteBuffer& packet) {
+  void SendPacketInternal(const CopyOnWriteBuffer& packet,
+      const PacketOptions& options) {
     if (dest_) {
-      dest_->SignalReadPacket(dest_, packet.data<char>(), packet.size(),
+      rtc::CopyOnWriteBuffer data(packet);
+      cricket::ApplyPacketOptions(
+          reinterpret_cast<uint8_t*>(data.data<char>()), data.size(),
+          options.packet_time_params, 0);
+      dest_->SignalReadPacket(dest_, data.data<char>(), data.size(),
                               CreatePacketTime(0), 0);
     }
   }
 
   AsyncInvoker invoker_;
   std::string debug_name_;
   FakePacketTransport* dest_ = nullptr;
   bool async_ = false;
   int async_delay_ms_ = 0;
   bool writable_ = false;
   bool receiving_ = false;
 };
 
 }  // namespace rtc
 
 #endif  // WEBRTC_P2P_BASE_FAKEPACKETTRANSPORT_H_