Remove all uses of GG_UINT32_C.

media/cast/audio_sender/audio_sender_unittest.cc

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
+#include <stdint.h>
+
 #include "base/bind.h"
 #include "base/bind_helpers.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/base/media.h"
 #include "media/cast/audio_sender/audio_sender.h"
 #include "media/cast/cast_config.h"
 #include "media/cast/cast_environment.h"
 #include "media/cast/rtcp/rtcp.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "media/cast/test/utility/audio_utility.h"
 #include "media/cast/transport/cast_transport_config.h"
 #include "media/cast/transport/cast_transport_sender_impl.h"
 #include "testing/gtest/include/gtest/gtest.h"
 
 namespace media {
 namespace cast {
 
-static const int64 kStartMillisecond = GG_INT64_C(12345678900000);
+static const int64 kStartMillisecond = INT64_C(12345678900000);
 
 class TestPacketSender : public transport::PacketSender {
  public:
   TestPacketSender() : number_of_rtp_packets_(0), number_of_rtcp_packets_(0) {}
 
   virtual bool SendPacket(const Packet& packet) OVERRIDE {
     if (Rtcp::IsRtcpPacket(&packet[0], packet.size())) {
       ++number_of_rtcp_packets_;
     } else {
       ++number_of_rtp_packets_;
@@ skipping 99 matching lines @@
   base::TimeDelta max_rtcp_timeout =
       base::TimeDelta::FromMilliseconds(1 + kDefaultRtcpIntervalMs * 3 / 2);
   testing_clock_->Advance(max_rtcp_timeout);
   task_runner_->RunTasks();
   EXPECT_GE(transport_.number_of_rtp_packets(), 1);
   EXPECT_EQ(transport_.number_of_rtcp_packets(), 1);
 }
 
 }  // namespace cast
 }  // namespace media