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

webrtc/modules/audio_coding/neteq/delay_manager.cc

Index: webrtc/modules/audio_coding/neteq/delay_manager.cc
diff --git a/webrtc/modules/audio_coding/neteq/delay_manager.cc b/webrtc/modules/audio_coding/neteq/delay_manager.cc
index a935561eff459bd386f414fc3a54105381718ea7..e7f76f616e9fc04c713467b944b1cf3f244c176c 100644
--- a/webrtc/modules/audio_coding/neteq/delay_manager.cc
+++ b/webrtc/modules/audio_coding/neteq/delay_manager.cc
@@ -22,7 +22,7 @@
 
 namespace webrtc {
 
-DelayManager::DelayManager(int max_packets_in_buffer,
+DelayManager::DelayManager(size_t max_packets_in_buffer,
                            DelayPeakDetector* peak_detector)
     : first_packet_received_(false),
       max_packets_in_buffer_(max_packets_in_buffer),
@@ -239,7 +239,8 @@ void DelayManager::LimitTargetLevel() {
   }
 
   // Shift to Q8, then 75%.;
-  int max_buffer_packets_q8 = (3 * (max_packets_in_buffer_ << 8)) / 4;
+  int max_buffer_packets_q8 =
+      static_cast<int>((3 * (max_packets_in_buffer_ << 8)) / 4);
   target_level_ = std::min(target_level_, max_buffer_packets_q8);
 
   // Sanity check, at least 1 packet (in Q8).
@@ -389,7 +390,8 @@ bool DelayManager::SetMinimumDelay(int delay_ms) {
   // |max_packets_in_buffer_|.
   if ((maximum_delay_ms_ > 0 && delay_ms > maximum_delay_ms_) ||
       (packet_len_ms_ > 0 &&
-          delay_ms > 3 * max_packets_in_buffer_ * packet_len_ms_ / 4)) {
+       delay_ms >
+           static_cast<int>(3 * max_packets_in_buffer_ * packet_len_ms_ / 4))) {
     return false;
   }
   minimum_delay_ms_ = delay_ms;