Convert channel counts to size_t.

webrtc/modules/audio_device/android/audio_record_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 25 matching lines @@
       init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
       start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
       stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
       enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
       enable_built_in_agc_(native_reg->GetMethodId("enableBuiltInAGC", "(Z)Z")),
       enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
 
 AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
 
 int AudioRecordJni::JavaAudioRecord::InitRecording(
-    int sample_rate, int channels) {
+    int sample_rate, size_t channels) {
   return audio_record_->CallIntMethod(init_recording_,
                                       static_cast<jint>(sample_rate),
                                       static_cast<jint>(channels));
 }
 
 bool AudioRecordJni::JavaAudioRecord::StartRecording() {
   return audio_record_->CallBooleanMethod(start_recording_);
 }
 
 bool AudioRecordJni::JavaAudioRecord::StopRecording() {
@@ skipping 121 matching lines @@
   return 0;
 }
 
 void AudioRecordJni::AttachAudioBuffer(AudioDeviceBuffer* audioBuffer) {
   ALOGD("AttachAudioBuffer");
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   audio_device_buffer_ = audioBuffer;
   const int sample_rate_hz = audio_parameters_.sample_rate();
   ALOGD("SetRecordingSampleRate(%d)", sample_rate_hz);
   audio_device_buffer_->SetRecordingSampleRate(sample_rate_hz);
-  const int channels = audio_parameters_.channels();
-  ALOGD("SetRecordingChannels(%d)", channels);
+  const size_t channels = audio_parameters_.channels();
+  ALOGD("SetRecordingChannels(%" PRIuS ")", channels);
   audio_device_buffer_->SetRecordingChannels(channels);
   total_delay_in_milliseconds_ =
       audio_manager_->GetDelayEstimateInMilliseconds();
   RTC_DCHECK_GT(total_delay_in_milliseconds_, 0);
   ALOGD("total_delay_in_milliseconds: %d", total_delay_in_milliseconds_);
 }
 
 int32_t AudioRecordJni::EnableBuiltInAEC(bool enable) {
   ALOGD("EnableBuiltInAEC%s", GetThreadInfo().c_str());
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 53 matching lines @@
   // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
   audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_,
                                    0,   // recDelayMs
                                    0);  // clockDrift
   if (audio_device_buffer_->DeliverRecordedData() == -1) {
     ALOGE("AudioDeviceBuffer::DeliverRecordedData failed!");
   }
 }
 
 }  // namespace webrtc