Convert channel counts to size_t.

webrtc/modules/audio_coding/neteq/tools/neteq_performance_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 91 matching lines @@
       assert(payload_len == kInputBlockSizeSamples * sizeof(int16_t));
     }
 
     // Get output audio, but don't do anything with it.
     static const int kMaxChannels = 1;
     static const size_t kMaxSamplesPerMs = 48000 / 1000;
     static const int kOutputBlockSizeMs = 10;
     static const size_t kOutDataLen =
         kOutputBlockSizeMs * kMaxSamplesPerMs * kMaxChannels;
     int16_t out_data[kOutDataLen];
-    int num_channels;
+    size_t num_channels;
     size_t samples_per_channel;
     int error = neteq->GetAudio(kOutDataLen, out_data, &samples_per_channel,
                                 &num_channels, NULL);
     if (error != NetEq::kOK)
       return -1;
 
     assert(samples_per_channel == static_cast<size_t>(kSampRateHz * 10 / 1000));
 
     time_now_ms += kOutputBlockSizeMs;
     if (time_now_ms >= runtime_ms / 2 && !drift_flipped) {
       // Apply negative drift second half of simulation.
       rtp_gen.set_drift_factor(-drift_factor);
       drift_flipped = true;
     }
   }
   int64_t end_time_ms = clock->TimeInMilliseconds();
   delete neteq;
   return end_time_ms - start_time_ms;
 }
 
 }  // namespace test
 }  // namespace webrtc