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

webrtc/modules/audio_coding/neteq/buffer_level_filter.cc

Index: webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
diff --git a/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc b/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
index 93f9a55b2c37374c29c570bdd9da5d150ff679b8..905479178d212c623a5a1ab054c51871b1f37799 100644
--- a/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
+++ b/webrtc/modules/audio_coding/neteq/buffer_level_filter.cc
@@ -23,16 +23,16 @@ void BufferLevelFilter::Reset() {
   level_factor_ = 253;
 }
 
-void BufferLevelFilter::Update(int buffer_size_packets,
+void BufferLevelFilter::Update(size_t buffer_size_packets,
                                int time_stretched_samples,
-                               int packet_len_samples) {
+                               size_t packet_len_samples) {
   // Filter:
   // |filtered_current_level_| = |level_factor_| * |filtered_current_level_| +
   //                            (1 - |level_factor_|) * |buffer_size_packets|
   // |level_factor_| and |filtered_current_level_| are in Q8.
   // |buffer_size_packets| is in Q0.
   filtered_current_level_ = ((level_factor_ * filtered_current_level_) >> 8) +
-      ((256 - level_factor_) * buffer_size_packets);
+      ((256 - level_factor_) * static_cast<int>(buffer_size_packets));
 
   // Account for time-scale operations (accelerate and pre-emptive expand).
   if (time_stretched_samples && packet_len_samples > 0) {
@@ -42,7 +42,7 @@ void BufferLevelFilter::Update(int buffer_size_packets,
     // Make sure that the filtered value remains non-negative.
     filtered_current_level_ = std::max(0,
         filtered_current_level_ -
-        (time_stretched_samples << 8) / packet_len_samples);
+        (time_stretched_samples << 8) / static_cast<int>(packet_len_samples));
   }
 }