Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h

Index: webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
diff --git a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
index d2b20e3b941c932cf736489b17cf4455154a9d32..a6aba3f720e8b9d8a4d8fbaf51f5144078c1c438 100644
--- a/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
+++ b/webrtc/modules/audio_coding/codecs/isac/audio_encoder_isac_t_impl.h
@@ -119,15 +119,16 @@ size_t AudioEncoderDecoderIsacT<T>::MaxEncodedBytes() const {
 }
 
 template <typename T>
-int AudioEncoderDecoderIsacT<T>::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderDecoderIsacT<T>::Num10MsFramesInNextPacket() const {
   CriticalSectionScoped cs(state_lock_.get());
   const int samples_in_next_packet = T::GetNewFrameLen(isac_state_);
-  return rtc::CheckedDivExact(samples_in_next_packet,
-                              rtc::CheckedDivExact(SampleRateHz(), 100));
+  return static_cast<size_t>(
+      rtc::CheckedDivExact(samples_in_next_packet,
+                           rtc::CheckedDivExact(SampleRateHz(), 100)));
 }
 
 template <typename T>
-int AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const {
+size_t AudioEncoderDecoderIsacT<T>::Max10MsFramesInAPacket() const {
   return 6;  // iSAC puts at most 60 ms in a packet.
 }
 
@@ -192,8 +193,7 @@ int AudioEncoderDecoderIsacT<T>::DecodeInternal(const uint8_t* encoded,
   }
   int16_t temp_type = 1;  // Default is speech.
   int ret =
-      T::DecodeInternal(isac_state_, encoded, static_cast<int16_t>(encoded_len),
-                        decoded, &temp_type);
+      T::DecodeInternal(isac_state_, encoded, encoded_len, decoded, &temp_type);
   *speech_type = ConvertSpeechType(temp_type);
   return ret;
 }
@@ -204,7 +204,8 @@ bool AudioEncoderDecoderIsacT<T>::HasDecodePlc() const {
 }
 
 template <typename T>
-int AudioEncoderDecoderIsacT<T>::DecodePlc(int num_frames, int16_t* decoded) {
+size_t AudioEncoderDecoderIsacT<T>::DecodePlc(size_t num_frames,
+                                              int16_t* decoded) {
   CriticalSectionScoped cs(state_lock_.get());
   return T::DecodePlc(isac_state_, decoded, num_frames);
 }
@@ -223,7 +224,7 @@ int AudioEncoderDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
                                                 uint32_t arrival_timestamp) {
   CriticalSectionScoped cs(state_lock_.get());
   return T::UpdateBwEstimate(
-      isac_state_, payload, static_cast<int32_t>(payload_len),
+      isac_state_, payload, payload_len,
       rtp_sequence_number, rtp_timestamp, arrival_timestamp);
 }