Convert some more things to size_t.

webrtc/modules/audio_device/ios/audio_device_ios.mm

 /*
  *  Copyright (c) 2012 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 68 matching lines @@
     // AVAudioSession object.
     AVAudioSession* session = [AVAudioSession sharedInstance];
     // Always get values when the audio session is active.
     ActivateAudioSession(session, true);
     CHECK(session.isInputAvailable) << "No input path is available!";
     // Get current hardware parameters.
     double sample_rate = (double)session.sampleRate;
     double io_buffer_duration = (double)session.IOBufferDuration;
     int output_channels = (int)session.outputNumberOfChannels;
     int input_channels = (int)session.inputNumberOfChannels;
-    int frames_per_buffer =
-        static_cast<int>(sample_rate * io_buffer_duration + 0.5);
+    size_t frames_per_buffer =
+        static_cast<size_t>(sample_rate * io_buffer_duration + 0.5);
     // Copy hardware parameters to output parameters.
     playout_parameters->reset(sample_rate, output_channels, frames_per_buffer);
     record_parameters->reset(sample_rate, input_channels, frames_per_buffer);
     // Add logging for debugging purposes.
     LOG(LS_INFO) << " sample rate: " << sample_rate;
     LOG(LS_INFO) << " IO buffer duration: " << io_buffer_duration;
     LOG(LS_INFO) << " frames_per_buffer: " << frames_per_buffer;
     LOG(LS_INFO) << " output channels: " << output_channels;
     LOG(LS_INFO) << " input channels: " << input_channels;
     LOG(LS_INFO) << " output latency: " << (double)session.outputLatency;
@@ skipping 944 matching lines @@
     //       Device Buffer?
     timespec t;
     t.tv_sec = 0;
     t.tv_nsec = 5 * 1000 * 1000;
     nanosleep(&t, nullptr);
   }
   return true;
 }
 
 }  // namespace webrtc