Convert channel counts to size_t.

webrtc/modules/audio_coding/test/target_delay_unittest.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 135 matching lines @@
                                       rtp_info_));
   }
 
   // Pull audio equivalent to the amount of audio in one RTP packet.
   void Pull() {
     AudioFrame frame;
     for (int k = 0; k < kNum10msPerFrame; ++k) {  // Pull one frame.
       ASSERT_EQ(0, acm_->PlayoutData10Ms(-1, &frame));
       // Had to use ASSERT_TRUE, ASSERT_EQ generated error.
       ASSERT_TRUE(kSampleRateHz == frame.sample_rate_hz_);
-      ASSERT_EQ(1, frame.num_channels_);
+      ASSERT_EQ(1u, frame.num_channels_);
       ASSERT_TRUE(kSampleRateHz / 100 == frame.samples_per_channel_);
     }
   }
 
   void Run(bool clean) {
     for (int n = 0; n < 10; ++n) {
       for (int m = 0; m < 5; ++m) {
         Push();
         Pull();
       }
@@ skipping 73 matching lines @@
 #define MAYBE_TargetDelayBufferMinMax DISABLED_TargetDelayBufferMinMax
 #else
 #define MAYBE_TargetDelayBufferMinMax TargetDelayBufferMinMax
 #endif
 TEST_F(TargetDelayTest, MAYBE_TargetDelayBufferMinMax) {
   TargetDelayBufferMinMax();
 }
 
 }  // namespace webrtc