Switch to standard integer types in media/.

media/cast/net/rtp/rtp_packetizer_unittest.cc

 // Copyright 2014 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 "media/cast/net/rtp/rtp_packetizer.h"
 
 #include <stdint.h>
 
 #include "base/memory/scoped_ptr.h"
 #include "base/test/simple_test_tick_clock.h"
 #include "media/cast/net/pacing/paced_sender.h"
 #include "media/cast/net/rtp/packet_storage.h"
 #include "media/cast/net/rtp/rtp_parser.h"
 #include "media/cast/test/fake_single_thread_task_runner.h"
 #include "testing/gmock/include/gmock/gmock.h"
 
 namespace media {
 namespace cast {
 
 namespace {
 static const int kPayload = 127;
-static const uint32 kTimestampMs = 10;
-static const uint16 kSeqNum = 33;
+static const uint32_t kTimestampMs = 10;
+static const uint16_t kSeqNum = 33;
 static const int kMaxPacketLength = 1500;
 static const int kSsrc = 0x12345;
 static const unsigned int kFrameSize = 5000;
 }
 
 class TestRtpPacketTransport : public PacketSender {
  public:
   explicit TestRtpPacketTransport(RtpPacketizerConfig config)
       : config_(config),
         sequence_number_(kSeqNum),
@@ skipping 26 matching lines @@
     if (rtp_header.packet_id != 0) {
       EXPECT_EQ(rtp_header.num_extensions, 0)
           << "Extensions only allowed on first packet of a frame";
     }
   }
 
   bool SendPacket(PacketRef packet, const base::Closure& cb) final {
     ++packets_sent_;
     RtpParser parser(kSsrc, kPayload);
     RtpCastHeader rtp_header;
-    const uint8* payload_data;
+    const uint8_t* payload_data;
     size_t payload_size;
     parser.ParsePacket(&packet->data[0], packet->data.size(), &rtp_header,
                        &payload_data, &payload_size);
     VerifyRtpHeader(rtp_header);
     ++sequence_number_;
     ++expected_packet_id_;
     return true;
   }
 
-  int64 GetBytesSent() final { return 0; }
+  int64_t GetBytesSent() final { return 0; }
 
   size_t number_of_packets_received() const { return packets_sent_; }
 
   void set_expected_number_of_packets(size_t expected_number_of_packets) {
     expected_number_of_packets_ = expected_number_of_packets;
   }
 
-  void set_rtp_timestamp(uint32 rtp_timestamp) {
+  void set_rtp_timestamp(uint32_t rtp_timestamp) {
     expected_rtp_timestamp_ = rtp_timestamp;
   }
 
   RtpPacketizerConfig config_;
-  uint32 sequence_number_;
+  uint32_t sequence_number_;
   size_t packets_sent_;
   size_t number_of_packets_;
   size_t expected_number_of_packets_;
   // Assuming packets arrive in sequence.
   int expected_packet_id_;
-  uint32 expected_frame_id_;
-  uint32 expected_rtp_timestamp_;
+  uint32_t expected_frame_id_;
+  uint32_t expected_rtp_timestamp_;
 
  private:
   DISALLOW_COPY_AND_ASSIGN(TestRtpPacketTransport);
 };
 
 class RtpPacketizerTest : public ::testing::Test {
  protected:
   RtpPacketizerTest()
       : task_runner_(new test::FakeSingleThreadTaskRunner(&testing_clock_)) {
     config_.sequence_number = kSeqNum;
@@ skipping 72 matching lines @@
   video_frame_.reference_time = testing_clock_.NowTicks();
   rtp_packetizer_->SendFrameAsPackets(video_frame_);
   RunTasks(33 + 1);
   EXPECT_EQ(expected_num_of_packets, rtp_packetizer_->send_packet_count());
   EXPECT_EQ(kFrameSize, rtp_packetizer_->send_octet_count());
   EXPECT_EQ(expected_num_of_packets, transport_->number_of_packets_received());
 }
 
 }  // namespace cast
 }  // namespace media