Convert channel counts to size_t.

webrtc/modules/audio_device/android/opensles_player.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 161 matching lines @@
   return -1;
 }
 
 void OpenSLESPlayer::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   ALOGD("AttachAudioBuffer");
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   audio_device_buffer_ = audioBuffer;
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetPlayoutSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetPlayoutSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetPlayoutChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetPlayoutChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetPlayoutChannels(channels);
   RTC_CHECK(audio_device_buffer_);
   AllocateDataBuffers();
 }
 
 SLDataFormat_PCM OpenSLESPlayer::CreatePCMConfiguration(
-    int channels,
+    size_t channels,
     int sample_rate,
     size_t bits_per_sample) {
   ALOGD("CreatePCMConfiguration");
   RTC_CHECK_EQ(bits_per_sample, SL_PCMSAMPLEFORMAT_FIXED_16);
   SLDataFormat_PCM format;
   format.formatType = SL_DATAFORMAT_PCM;
   format.numChannels = static_cast<SLuint32>(channels);
   // Note that, the unit of sample rate is actually in milliHertz and not Hertz.
   switch (sample_rate) {
     case 8000:
@@ skipping 257 matching lines @@
   RTC_DCHECK(player_);
   SLuint32 state;
   SLresult err = (*player_)->GetPlayState(player_, &state);
   if (SL_RESULT_SUCCESS != err) {
     ALOGE("GetPlayState failed: %d", err);
   }
   return state;
 }
 
 }  // namespace webrtc