Convert channel counts to size_t.

webrtc/modules/audio_coding/neteq/tools/neteq_quality_test.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 192 matching lines @@
     iter ++;
   }
   return x;
 }
 
 NetEqQualityTest::NetEqQualityTest(int block_duration_ms,
                                    int in_sampling_khz,
                                    int out_sampling_khz,
                                    NetEqDecoder decoder_type)
     : decoder_type_(decoder_type),
-      channels_(FLAGS_channels),
+      channels_(static_cast<size_t>(FLAGS_channels)),
       decoded_time_ms_(0),
       decodable_time_ms_(0),
       drift_factor_(FLAGS_drift_factor),
       packet_loss_rate_(FLAGS_packet_loss_rate),
       block_duration_ms_(block_duration_ms),
       in_sampling_khz_(in_sampling_khz),
       out_sampling_khz_(out_sampling_khz),
       in_size_samples_(
           static_cast<size_t>(in_sampling_khz_ * block_duration_ms_)),
       out_size_samples_(static_cast<size_t>(out_sampling_khz_ * kOutputSizeMs)),
@@ skipping 163 matching lines @@
       Log() << "was sent.";
     } else {
       Log() << "was lost.";
     }
   }
   Log() << std::endl;
   return packet_input_time_ms;
 }
 
 int NetEqQualityTest::DecodeBlock() {
-  int channels;
+  size_t channels;
   size_t samples;
   int ret = neteq_->GetAudio(out_size_samples_ * channels_, &out_data_[0],
                              &samples, &channels, NULL);
 
   if (ret != NetEq::kOK) {
     return -1;
   } else {
     assert(channels == channels_);
     assert(samples == static_cast<size_t>(kOutputSizeMs * out_sampling_khz_));
     RTC_CHECK(output_->WriteArray(out_data_.get(), samples * channels));
@@ skipping 20 matching lines @@
     }
   }
   Log() << "Average bit rate was "
         << 8.0f * total_payload_size_bytes_ / FLAGS_runtime_ms
         << " kbps"
         << std::endl;
 }
 
 }  // namespace test
 }  // namespace webrtc