Removed vie_defines.h

webrtc/modules/rtp_rtcp/source/rtp_packet_history_unittest.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.
  *
  * This file includes unit tests for the RTPPacketHistory.
  */
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_packet_history.h"
 #include "webrtc/system_wrappers/include/clock.h"
-#include "webrtc/video_engine/vie_defines.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
 
 class RtpPacketHistoryTest : public ::testing::Test {
  protected:
   RtpPacketHistoryTest()
      : fake_clock_(123456),
        hist_(new RTPPacketHistory(&fake_clock_)) {
   }
   ~RtpPacketHistoryTest() {
     delete hist_;
   }
-  
+
   SimulatedClock fake_clock_;
   RTPPacketHistory* hist_;
   enum {kPayload = 127};
   enum {kSsrc = 12345678};
   enum {kSeqNum = 88};
   enum {kTimestamp = 127};
   enum {kMaxPacketLength = 1500};
   uint8_t packet_[kMaxPacketLength];
   uint8_t packet_out_[kMaxPacketLength];
 
   void CreateRtpPacket(uint16_t seq_num, uint32_t ssrc, uint8_t payload,
       uint32_t timestamp, uint8_t* array, size_t* cur_pos) {
     array[(*cur_pos)++] = 0x80;
     array[(*cur_pos)++] = payload;
     array[(*cur_pos)++] = seq_num >> 8;
     array[(*cur_pos)++] = seq_num;
     array[(*cur_pos)++] = timestamp >> 24;
     array[(*cur_pos)++] = timestamp >> 16;
     array[(*cur_pos)++] = timestamp >> 8;
     array[(*cur_pos)++] = timestamp;
     array[(*cur_pos)++] = ssrc >> 24;
     array[(*cur_pos)++] = ssrc >> 16;
     array[(*cur_pos)++] = ssrc >> 8;
     array[(*cur_pos)++] = ssrc;
-  } 
+  }
 };
 
 TEST_F(RtpPacketHistoryTest, SetStoreStatus) {
   EXPECT_FALSE(hist_->StorePackets());
   hist_->SetStorePacketsStatus(true, 10);
   EXPECT_TRUE(hist_->StorePackets());
   hist_->SetStorePacketsStatus(false, 0);
   EXPECT_FALSE(hist_->StorePackets());
 }
 
@@ skipping 193 matching lines @@
 
   // Retransmit last 16 packets.
   for (int i = 4; i < 20; ++i) {
     len = kMaxPacketLength;
     EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum + i, 100, false, packet_,
                                                &len, &time));
   }
 }
 
 TEST_F(RtpPacketHistoryTest, FullExpansion) {
+  static const int kSendSidePacketHistorySize = 600;
   hist_->SetStorePacketsStatus(true, kSendSidePacketHistorySize);
   size_t len;
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   int64_t time;
   for (size_t i = 0; i < kMaxHistoryCapacity + 1; ++i) {
     len = 0;
     CreateRtpPacket(kSeqNum + i, kSsrc, kPayload, kTimestamp, packet_, &len);
     EXPECT_EQ(0, hist_->PutRTPPacket(packet_, len, capture_time_ms,
                                      kAllowRetransmission));
   }
 
   fake_clock_.AdvanceTimeMilliseconds(100);
 
   // Retransmit all packets currently in buffer.
   for (size_t i = 1; i < kMaxHistoryCapacity + 1; ++i) {
     len = kMaxPacketLength;
     EXPECT_TRUE(hist_->GetPacketAndSetSendTime(kSeqNum + i, 100, false, packet_,
                                                &len, &time));
   }
 }
 
 }  // namespace webrtc