Convert channel counts to size_t.

webrtc/modules/audio_processing/test/debug_dump_test.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 309 matching lines @@
   ASSERT_TRUE(apm_.get());
 
   EXPECT_NOERR(apm_->gain_control()->set_stream_analog_level(msg.level()));
   EXPECT_NOERR(apm_->set_stream_delay_ms(msg.delay()));
   apm_->echo_cancellation()->set_stream_drift_samples(msg.drift());
   if (msg.has_keypress())
     apm_->set_stream_key_pressed(msg.keypress());
   else
     apm_->set_stream_key_pressed(true);
 
-  ASSERT_EQ(input_config_.num_channels(), msg.input_channel_size());
+  ASSERT_EQ(input_config_.num_channels(),
+            static_cast<size_t>(msg.input_channel_size()));
   ASSERT_EQ(input_config_.num_frames() * sizeof(float),
             msg.input_channel(0).size());
 
   for (int i = 0; i < msg.input_channel_size(); ++i) {
      memcpy(input_->channels()[i], msg.input_channel(i).data(),
             msg.input_channel(i).size());
   }
 
   ASSERT_EQ(AudioProcessing::kNoError,
             apm_->ProcessStream(input_->channels(), input_config_,
                                 output_config_, output_->channels()));
 
   // Check that output of APM is bit-exact to the output in the dump.
-  ASSERT_EQ(output_config_.num_channels(), msg.output_channel_size());
+  ASSERT_EQ(output_config_.num_channels(),
+            static_cast<size_t>(msg.output_channel_size()));
   ASSERT_EQ(output_config_.num_frames() * sizeof(float),
             msg.output_channel(0).size());
   for (int i = 0; i < msg.output_channel_size(); ++i) {
     ASSERT_EQ(0, memcmp(output_->channels()[i], msg.output_channel(i).data(),
                         msg.output_channel(i).size()));
   }
 }
 
 void DebugDumpTest::OnReverseStreamEvent(const audioproc::ReverseStream& msg) {
   // APM should have been created.
   ASSERT_TRUE(apm_.get());
 
   ASSERT_GT(msg.channel_size(), 0);
-  ASSERT_EQ(reverse_config_.num_channels(), msg.channel_size());
+  ASSERT_EQ(reverse_config_.num_channels(),
+            static_cast<size_t>(msg.channel_size()));
   ASSERT_EQ(reverse_config_.num_frames() * sizeof(float),
             msg.channel(0).size());
 
   for (int i = 0; i < msg.channel_size(); ++i) {
      memcpy(reverse_->channels()[i], msg.channel(i).data(),
             msg.channel(i).size());
   }
 
   ASSERT_EQ(AudioProcessing::kNoError,
             apm_->ProcessReverseStream(reverse_->channels(),
@@ skipping 231 matching lines @@
   config.Set<ExperimentalNs>(new ExperimentalNs(true));
   DebugDumpGenerator generator(config);
   generator.StartRecording();
   generator.Process(100);
   generator.StopRecording();
   VerifyDebugDump(generator.dump_file_name());
 }
 
 }  // namespace test
 }  // namespace webrtc