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

webrtc/common_audio/vad/vad_sp_unittest.cc

Index: webrtc/common_audio/vad/vad_sp_unittest.cc
diff --git a/webrtc/common_audio/vad/vad_sp_unittest.cc b/webrtc/common_audio/vad/vad_sp_unittest.cc
index 9127bf329901faae94941a3a18ec03bb83f8f18f..6d5e2a646b636d7085a346a3eabd28012babb9bf 100644
--- a/webrtc/common_audio/vad/vad_sp_unittest.cc
+++ b/webrtc/common_audio/vad/vad_sp_unittest.cc
@@ -23,7 +23,7 @@ namespace {
 
 TEST_F(VadTest, vad_sp) {
   VadInstT* self = reinterpret_cast<VadInstT*>(malloc(sizeof(VadInstT)));
-  const int kMaxFrameLenSp = 960;  // Maximum frame length in this unittest.
+  const size_t kMaxFrameLenSp = 960;  // Maximum frame length in this unittest.
   int16_t zeros[kMaxFrameLenSp] = { 0 };
   int32_t state[2] = { 0 };
   int16_t data_in[kMaxFrameLenSp];
@@ -40,14 +40,14 @@ TEST_F(VadTest, vad_sp) {
 
   // Construct a speech signal that will trigger the VAD in all modes. It is
   // known that (i * i) will wrap around, but that doesn't matter in this case.
-  for (int16_t i = 0; i < kMaxFrameLenSp; ++i) {
+  for (size_t i = 0; i < kMaxFrameLenSp; ++i) {
     data_in[i] = static_cast<int16_t>(i * i);
   }
   // Input values all zeros, expect all zeros out.
   WebRtcVad_Downsampling(zeros, data_out, state, kMaxFrameLenSp);
   EXPECT_EQ(0, state[0]);
   EXPECT_EQ(0, state[1]);
-  for (int16_t i = 0; i < kMaxFrameLenSp / 2; ++i) {
+  for (size_t i = 0; i < kMaxFrameLenSp / 2; ++i) {
     EXPECT_EQ(0, data_out[i]);
   }
   // Make a simple non-zero data test.