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

webrtc/common_audio/signal_processing/splitting_filter.c

Index: webrtc/common_audio/signal_processing/splitting_filter.c
diff --git a/webrtc/common_audio/signal_processing/splitting_filter.c b/webrtc/common_audio/signal_processing/splitting_filter.c
index 7ae281c2ee43904bbdfad81fb43c86e07dfdcb76..36fcf355ecc3eb63a55c261058a6711471594b33 100644
--- a/webrtc/common_audio/signal_processing/splitting_filter.c
+++ b/webrtc/common_audio/signal_processing/splitting_filter.c
@@ -45,7 +45,7 @@ static const uint16_t WebRtcSpl_kAllPassFilter2[3] = {21333, 49062, 63010};
 //                            |data_length|
 //
 
-void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
+void WebRtcSpl_AllPassQMF(int32_t* in_data, size_t data_length,
                           int32_t* out_data, const uint16_t* filter_coefficients,
                           int32_t* filter_state)
 {
@@ -65,7 +65,7 @@ void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
     // filter operation takes the |in_data| (which is the output from the previous cascade
     // filter) and store the output in |out_data|.
     // Note that the input vector values are changed during the process.
-    int k;
+    size_t k;
     int32_t diff;
     // First all-pass cascade; filter from in_data to out_data.
 
@@ -124,18 +124,18 @@ void WebRtcSpl_AllPassQMF(int32_t* in_data, int data_length,
     filter_state[5] = out_data[data_length - 1]; // y[N-1], becomes y[-1] next time
 }
 
-void WebRtcSpl_AnalysisQMF(const int16_t* in_data, int in_data_length,
+void WebRtcSpl_AnalysisQMF(const int16_t* in_data, size_t in_data_length,
                            int16_t* low_band, int16_t* high_band,
                            int32_t* filter_state1, int32_t* filter_state2)
 {
-    int16_t i;
+    size_t i;
     int16_t k;
     int32_t tmp;
     int32_t half_in1[kMaxBandFrameLength];
     int32_t half_in2[kMaxBandFrameLength];
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
-    const int band_length = in_data_length / 2;
+    const size_t band_length = in_data_length / 2;
     assert(in_data_length % 2 == 0);
     assert(band_length <= kMaxBandFrameLength);
 
@@ -165,7 +165,7 @@ void WebRtcSpl_AnalysisQMF(const int16_t* in_data, int in_data_length,
 }
 
 void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
-                            int band_length, int16_t* out_data,
+                            size_t band_length, int16_t* out_data,
                             int32_t* filter_state1, int32_t* filter_state2)
 {
     int32_t tmp;
@@ -173,7 +173,7 @@ void WebRtcSpl_SynthesisQMF(const int16_t* low_band, const int16_t* high_band,
     int32_t half_in2[kMaxBandFrameLength];
     int32_t filter1[kMaxBandFrameLength];
     int32_t filter2[kMaxBandFrameLength];
-    int16_t i;
+    size_t i;
     int16_t k;
     assert(band_length <= kMaxBandFrameLength);