Update audio code to use size_t more correctly, talk/ portion.

talk/app/webrtc/test/fakeaudiocapturemodule.cc

Index: talk/app/webrtc/test/fakeaudiocapturemodule.cc
diff --git a/talk/app/webrtc/test/fakeaudiocapturemodule.cc b/talk/app/webrtc/test/fakeaudiocapturemodule.cc
index 47f17a1739abe566a38bad9d14ebdfa24b0507d2..cbd5133bd23a6076113473bc8ca8e838fe0c29e7 100644
--- a/talk/app/webrtc/test/fakeaudiocapturemodule.cc
+++ b/talk/app/webrtc/test/fakeaudiocapturemodule.cc
@@ -623,9 +623,9 @@ bool FakeAudioCaptureModule::Initialize() {
 
 void FakeAudioCaptureModule::SetSendBuffer(int value) {
   Sample* buffer_ptr = reinterpret_cast<Sample*>(send_buffer_);
-  const int buffer_size_in_samples =
+  const size_t buffer_size_in_samples =
       sizeof(send_buffer_) / kNumberBytesPerSample;
-  for (int i = 0; i < buffer_size_in_samples; ++i) {
+  for (size_t i = 0; i < buffer_size_in_samples; ++i) {
     buffer_ptr[i] = value;
   }
 }
@@ -636,9 +636,9 @@ void FakeAudioCaptureModule::ResetRecBuffer() {
 
 bool FakeAudioCaptureModule::CheckRecBuffer(int value) {
   const Sample* buffer_ptr = reinterpret_cast<const Sample*>(rec_buffer_);
-  const int buffer_size_in_samples =
+  const size_t buffer_size_in_samples =
       sizeof(rec_buffer_) / kNumberBytesPerSample;
-  for (int i = 0; i < buffer_size_in_samples; ++i) {
+  for (size_t i = 0; i < buffer_size_in_samples; ++i) {
     if (buffer_ptr[i] >= value) return true;
   }
   return false;
@@ -703,7 +703,7 @@ void FakeAudioCaptureModule::ReceiveFrameP() {
       return;
     }
     ResetRecBuffer();
-    uint32_t nSamplesOut = 0;
+    size_t nSamplesOut = 0;
     int64_t elapsed_time_ms = 0;
     int64_t ntp_time_ms = 0;
     if (audio_callback_->NeedMorePlayData(kNumberSamples, kNumberBytesPerSample,