Update a ton of audio code to use size_t more correctly and in general reduce

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.
  */
 
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/audio_track_jni.h"
 
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 
 #define TAG "AudioTrackJni"
 #define ALOGV(...) __android_log_print(ANDROID_LOG_VERBOSE, TAG, __VA_ARGS__)
 #define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
 #define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
 #define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
 
 namespace webrtc {
 
@@ skipping 182 matching lines @@
 }
 
 void AudioTrackJni::OnCacheDirectBufferAddress(
     JNIEnv* env, jobject byte_buffer) {
   ALOGD("OnCacheDirectBufferAddress");
   DCHECK(thread_checker_.CalledOnValidThread());
   direct_buffer_address_ =
       env->GetDirectBufferAddress(byte_buffer);
   jlong capacity = env->GetDirectBufferCapacity(byte_buffer);
   ALOGD("direct buffer capacity: %lld", capacity);
-  direct_buffer_capacity_in_bytes_ = static_cast<int> (capacity);
+  direct_buffer_capacity_in_bytes_ = static_cast<size_t>(capacity);
   frames_per_buffer_ = direct_buffer_capacity_in_bytes_ / kBytesPerFrame;
-  ALOGD("frames_per_buffer: %d", frames_per_buffer_);
+  ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
 }
 
 void JNICALL AudioTrackJni::GetPlayoutData(
   JNIEnv* env, jobject obj, jint length, jlong nativeAudioTrack) {
   webrtc::AudioTrackJni* this_object =
       reinterpret_cast<webrtc::AudioTrackJni*> (nativeAudioTrack);
-  this_object->OnGetPlayoutData(length);
+  this_object->OnGetPlayoutData(static_cast<size_t>(length));
 }
 
 // This method is called on a high-priority thread from Java. The name of
 // the thread is 'AudioRecordTrack'.
-void AudioTrackJni::OnGetPlayoutData(int length) {
+void AudioTrackJni::OnGetPlayoutData(size_t length) {
   DCHECK(thread_checker_java_.CalledOnValidThread());
   DCHECK_EQ(frames_per_buffer_, length / kBytesPerFrame);
   if (!audio_device_buffer_) {
     ALOGE("AttachAudioBuffer has not been called!");
     return;
   }
   // Pull decoded data (in 16-bit PCM format) from jitter buffer.
   int samples = audio_device_buffer_->RequestPlayoutData(frames_per_buffer_);
   if (samples <= 0) {
     ALOGE("AudioDeviceBuffer::RequestPlayoutData failed!");
     return;
   }
-  DCHECK_EQ(samples, frames_per_buffer_);
+  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_);
   DCHECK_EQ(length, kBytesPerFrame * samples);
 }
 
 }  // namespace webrtc