Convert channel counts to size_t.

webrtc/modules/audio_processing/test/audio_file_processor.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 114 matching lines @@
   reverse_config_ =
       StreamConfig(reverse_sample_rate, msg.num_reverse_channels());
 
   const ProcessingConfig config = {
       {input_config_, output_config_, reverse_config_, reverse_config_}};
   RTC_CHECK_EQ(kNoErr, ap_->Initialize(config));
 }
 
 void AecDumpFileProcessor::HandleMessage(const Stream& msg) {
   RTC_CHECK(!msg.has_input_data());
-  RTC_CHECK_EQ(in_buf_->num_channels(), msg.input_channel_size());
+  RTC_CHECK_EQ(in_buf_->num_channels(),
+               static_cast<size_t>(msg.input_channel_size()));
 
   for (int i = 0; i < msg.input_channel_size(); ++i) {
     RTC_CHECK_EQ(in_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
                  msg.input_channel(i).size());
     std::memcpy(in_buf_->channels()[i], msg.input_channel(i).data(),
                 msg.input_channel(i).size());
   }
   {
     const auto st = ScopedTimer(mutable_proc_time());
     RTC_CHECK_EQ(kNoErr, ap_->set_stream_delay_ms(msg.delay()));
     ap_->echo_cancellation()->set_stream_drift_samples(msg.drift());
     if (msg.has_keypress()) {
       ap_->set_stream_key_pressed(msg.keypress());
     }
     RTC_CHECK_EQ(kNoErr,
                  ap_->ProcessStream(in_buf_->channels(), input_config_,
                                     output_config_, out_buf_.channels()));
   }
 
   buffer_writer_.Write(out_buf_);
 }
 
 void AecDumpFileProcessor::HandleMessage(const ReverseStream& msg) {
   RTC_CHECK(!msg.has_data());
-  RTC_CHECK_EQ(reverse_buf_->num_channels(), msg.channel_size());
+  RTC_CHECK_EQ(reverse_buf_->num_channels(),
+               static_cast<size_t>(msg.channel_size()));
 
   for (int i = 0; i < msg.channel_size(); ++i) {
     RTC_CHECK_EQ(reverse_buf_->num_frames() * sizeof(*in_buf_->channels()[i]),
                  msg.channel(i).size());
     std::memcpy(reverse_buf_->channels()[i], msg.channel(i).data(),
                 msg.channel(i).size());
   }
   {
     const auto st = ScopedTimer(mutable_proc_time());
     // TODO(ajm): This currently discards the processed output, which is needed
     // for e.g. intelligibility enhancement.
     RTC_CHECK_EQ(kNoErr, ap_->ProcessReverseStream(
                              reverse_buf_->channels(), reverse_config_,
                              reverse_config_, reverse_buf_->channels()));
   }
 }
 
 }  // namespace webrtc