More iOS compile fixes.

webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc

 /*
  *  Copyright (c) 2015 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 735 matching lines @@
   mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
   // SetMaxPlayoutVolume();
   EXPECT_EQ(0, audio_device()->RegisterAudioCallback(&mock));
   StartRecording();
   StartPlayout();
   test_is_done_->Wait(
       std::max(kTestTimeOutInMilliseconds, 1000 * kFullDuplexTimeInSec));
   StopPlayout();
   StopRecording();
-  EXPECT_LE(fifo_audio_stream->average_size(), 10);
-  EXPECT_LE(fifo_audio_stream->largest_size(), 20);
+  EXPECT_LE(fifo_audio_stream->average_size(), 10u);
+  EXPECT_LE(fifo_audio_stream->largest_size(), 20u);
 }
 
 // Measures loopback latency and reports the min, max and average values for
 // a full duplex audio session.
 // The latency is measured like so:
 // - Insert impulses periodically on the output side.
 // - Detect the impulses on the input side.
 // - Measure the time difference between the transmit time and receive time.
 // - Store time differences in a vector and calculate min, max and average.
 // This test requires a special hardware called Audio Loopback Dongle.
@@ skipping 10 matching lines @@
   // SetMaxPlayoutVolume();
   // DisableBuiltInAECIfAvailable();
   StartRecording();
   StartPlayout();
   test_is_done_->Wait(
       std::max(kTestTimeOutInMilliseconds, 1000 * kMeasureLatencyTimeInSec));
   StopPlayout();
   StopRecording();
   // Verify that the correct number of transmitted impulses are detected.
   EXPECT_EQ(latency_audio_stream->num_latency_values(),
-            kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1);
+            static_cast<size_t>(
+                kImpulseFrequencyInHz * kMeasureLatencyTimeInSec - 1));
   latency_audio_stream->PrintResults();
 }
 
 }  // namespace webrtc