Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.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.
  */
 
@@ skipping 24 matching lines @@
   AudioEncoderCopyRedTest()
       : timestamp_(4711),
         sample_rate_hz_(16000),
         num_audio_samples_10ms(sample_rate_hz_ / 100),
         red_payload_type_(200) {
     AudioEncoderCopyRed::Config config;
     config.payload_type = red_payload_type_;
     config.speech_encoder = &mock_encoder_;
     red_.reset(new AudioEncoderCopyRed(config));
     memset(audio_, 0, sizeof(audio_));
-    EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1));
+    EXPECT_CALL(mock_encoder_, NumChannels()).WillRepeatedly(Return(1U));
     EXPECT_CALL(mock_encoder_, SampleRateHz())
         .WillRepeatedly(Return(sample_rate_hz_));
     EXPECT_CALL(mock_encoder_, MaxEncodedBytes())
         .WillRepeatedly(Return(kMockMaxEncodedBytes));
     encoded_.resize(red_->MaxEncodedBytes(), 0);
   }
 
   void TearDown() override {
     red_.reset();
     // Don't expect the red_ object to delete the AudioEncoder object. But it
@@ skipping 47 matching lines @@
 
 TEST_F(AudioEncoderCopyRedTest, CreateAndDestroy) {
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckSampleRatePropagation) {
   EXPECT_CALL(mock_encoder_, SampleRateHz()).WillOnce(Return(17));
   EXPECT_EQ(17, red_->SampleRateHz());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckNumChannelsPropagation) {
-  EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17));
-  EXPECT_EQ(17, red_->NumChannels());
+  EXPECT_CALL(mock_encoder_, NumChannels()).WillOnce(Return(17U));
+  EXPECT_EQ(17U, red_->NumChannels());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckFrameSizePropagation) {
   EXPECT_CALL(mock_encoder_, Num10MsFramesInNextPacket()).WillOnce(Return(17U));
   EXPECT_EQ(17U, red_->Num10MsFramesInNextPacket());
 }
 
 TEST_F(AudioEncoderCopyRedTest, CheckMaxFrameSizePropagation) {
   EXPECT_CALL(mock_encoder_, Max10MsFramesInAPacket()).WillOnce(Return(17U));
   EXPECT_EQ(17U, red_->Max10MsFramesInAPacket());
@@ skipping 202 matching lines @@
   config.speech_encoder = NULL;
   EXPECT_DEATH(red = new AudioEncoderCopyRed(config),
                "Speech encoder not provided.");
   // The delete operation is needed to avoid leak reports from memcheck.
   delete red;
 }
 
 #endif  // GTEST_HAS_DEATH_TEST && !defined(WEBRTC_ANDROID)
 
 }  // namespace webrtc