Update audio code to use size_t more correctly, webrtc/modules/audio_device/

webrtc/modules/audio_device/android/audio_record_jni.cc

 /*
  *  Copyright (c) 2013 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.
  */
 
@@ skipping 108 matching lines @@
   int frames_per_buffer = j_audio_record_->InitRecording(
       audio_parameters_.sample_rate(), audio_parameters_.channels());
   if (frames_per_buffer < 0) {
     ALOGE("InitRecording failed!");
     return -1;
   }
   frames_per_buffer_ = frames_per_buffer;
   ALOGD("frames_per_buffer: %d", frames_per_buffer_);
   CHECK_EQ(direct_buffer_capacity_in_bytes_,
            frames_per_buffer_ * kBytesPerFrame);
-  CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
+  CHECK_EQ(frames_per_buffer_,
+           static_cast<int>(audio_parameters_.frames_per_10ms_buffer()));
   initialized_ = true;
   return 0;
 }
 
 int32_t AudioRecordJni::StartRecording() {
   ALOGD("StartRecording%s", GetThreadInfo().c_str());
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(initialized_);
   DCHECK(!recording_);
   if (!j_audio_record_->StartRecording()) {
@@ skipping 85 matching lines @@
   // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
   audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_,
                                    0,   // recDelayMs
                                    0);  // clockDrift
   if (audio_device_buffer_->DeliverRecordedData() == -1) {
     ALOGE("AudioDeviceBuffer::DeliverRecordedData failed!");
   }
 }
 
 }  // namespace webrtc