Convert channel counts to size_t.

webrtc/modules/audio_device/android/audio_track_jni.cc

 /*
  *  Copyright (c) 2013 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 184 matching lines @@
 }
 
 // TODO(henrika): possibly add stereo support.
 void AudioTrackJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   ALOGD("AttachAudioBuffer%s", GetThreadInfo().c_str());
   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);
 }
 
 void JNICALL AudioTrackJni::CacheDirectBufferAddress(
     JNIEnv* env, jobject obj, jobject byte_buffer, jlong nativeAudioTrack) {
   webrtc::AudioTrackJni* this_object =
       reinterpret_cast<webrtc::AudioTrackJni*> (nativeAudioTrack);
   this_object->OnCacheDirectBufferAddress(env, byte_buffer);
 }
 
@@ skipping 34 matching lines @@
     return;
   }
   RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_);
   // Copy decoded data into common byte buffer to ensure that it can be
   // written to the Java based audio track.
   samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
   RTC_DCHECK_EQ(length, kBytesPerFrame * samples);
 }
 
 }  // namespace webrtc