Update includes for webrtc/{base => rtc_base} rename (1/3)

webrtc/modules/audio_coding/acm2/acm_send_test.cc

 /*
  *  Copyright (c) 2014 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.
  */
 
 #include "webrtc/modules/audio_coding/acm2/acm_send_test.h"
 
 #include <assert.h>
 #include <stdio.h>
 #include <string.h>
 
 #include "webrtc/api/audio_codecs/audio_encoder.h"
-#include "webrtc/base/checks.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/audio_coding/neteq/tools/input_audio_file.h"
 #include "webrtc/modules/audio_coding/neteq/tools/packet.h"
+#include "webrtc/rtc_base/checks.h"
 #include "webrtc/test/gtest.h"
 
 namespace webrtc {
 namespace test {
 
 AcmSendTestOldApi::AcmSendTestOldApi(InputAudioFile* audio_source,
                                      int source_rate_hz,
                                      int test_duration_ms)
     : clock_(0),
       acm_(webrtc::AudioCodingModule::Create(0, &clock_)),
@@ skipping 119 matching lines @@
          last_payload_vec_.size());
   std::unique_ptr<Packet> packet(
       new Packet(packet_memory, allocated_bytes, clock_.TimeInMilliseconds()));
   RTC_DCHECK(packet);
   RTC_DCHECK(packet->valid_header());
   return packet;
 }
 
 }  // namespace test
 }  // namespace webrtc