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

webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc

Index: webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
index 0c246c34bdb6a825ea1ca9014254157a2e1cf890..4ca6fe94a8fe6143cd23512230128c6e0338ca58 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.cc
@@ -23,10 +23,10 @@ bool AudioEncoderPcm16B::Config::IsOk() const {
   return AudioEncoderPcm::Config::IsOk();
 }
 
-int16_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
-                                       size_t input_len,
-                                       uint8_t* encoded) {
-  return WebRtcPcm16b_Encode(audio, static_cast<int16_t>(input_len), encoded);
+size_t AudioEncoderPcm16B::EncodeCall(const int16_t* audio,
+                                      size_t input_len,
+                                      uint8_t* encoded) {
+  return WebRtcPcm16b_Encode(audio, input_len, encoded);
 }
 
 int AudioEncoderPcm16B::BytesPerSample() const {