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

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.
  */
 
 #include "webrtc/modules/audio_device/android/audio_record_jni.h"
 
 #include <android/log.h>
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
+#include "webrtc/base/format_macros.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 
 #define TAG "AudioRecordJni"
 #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 88 matching lines @@
   ALOGD("InitRecording%s", GetThreadInfo().c_str());
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!initialized_);
   DCHECK(!recording_);
   int frames_per_buffer = j_audio_record_->InitRecording(
       audio_parameters_.sample_rate(), audio_parameters_.channels());
   if (frames_per_buffer < 0) {
     ALOGE("InitRecording failed!");
     return -1;
   }
-  frames_per_buffer_ = frames_per_buffer;
-  ALOGD("frames_per_buffer: %d", frames_per_buffer_);
+  frames_per_buffer_ = static_cast<size_t>(frames_per_buffer);
+  ALOGD("frames_per_buffer: %" PRIuS, frames_per_buffer_);
   CHECK_EQ(direct_buffer_capacity_in_bytes_,
            frames_per_buffer_ * kBytesPerFrame);
   CHECK_EQ(frames_per_buffer_, audio_parameters_.frames_per_10ms_buffer());
   initialized_ = true;
   return 0;
 }
 
 int32_t AudioRecordJni::StartRecording() {
   ALOGD("StartRecording%s", GetThreadInfo().c_str());
   DCHECK(thread_checker_.CalledOnValidThread());
@@ skipping 56 matching lines @@
 
 void AudioRecordJni::OnCacheDirectBufferAddress(
     JNIEnv* env, jobject byte_buffer) {
   ALOGD("OnCacheDirectBufferAddress");
   DCHECK(thread_checker_.CalledOnValidThread());
   DCHECK(!direct_buffer_address_);
   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);
 }
 
 void JNICALL AudioRecordJni::DataIsRecorded(
   JNIEnv* env, jobject obj, jint length, jlong nativeAudioRecord) {
   webrtc::AudioRecordJni* this_object =
       reinterpret_cast<webrtc::AudioRecordJni*> (nativeAudioRecord);
   this_object->OnDataIsRecorded(length);
 }
 
 // This method is called on a high-priority thread from Java. The name of
@@ skipping 11 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