Update audio code to use size_t more correctly,

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

 /*
  *  Copyright (c) 2014 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 101 matching lines @@
 int AudioEncoderDecoderIsacT<T>::NumChannels() const {
   return 1;
 }
 
 template <typename T>
 size_t AudioEncoderDecoderIsacT<T>::MaxEncodedBytes() const {
   return kSufficientEncodeBufferSizeBytes;
 }
 
 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.
 }
 
 template <typename T>
 int AudioEncoderDecoderIsacT<T>::GetTargetBitrate() const {
   return target_bitrate_bps_;
 }
 
 template <typename T>
 AudioEncoder::EncodedInfo AudioEncoderDecoderIsacT<T>::EncodeInternal(
@@ skipping 44 matching lines @@
   if (sample_rate_hz == 48000)
     sample_rate_hz = 32000;
   CHECK(sample_rate_hz == 16000 || sample_rate_hz == 32000)
       << "Unsupported sample rate " << sample_rate_hz;
   if (sample_rate_hz != decoder_sample_rate_hz_) {
     CHECK_EQ(0, T::SetDecSampRate(isac_state_, sample_rate_hz));
     decoder_sample_rate_hz_ = sample_rate_hz;
   }
   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;
 }
 
 template <typename T>
 bool AudioEncoderDecoderIsacT<T>::HasDecodePlc() const {
   return false;
 }
 
 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);
 }
 
 template <typename T>
 int AudioEncoderDecoderIsacT<T>::Init() {
   CriticalSectionScoped cs(state_lock_.get());
   return T::DecoderInit(isac_state_);
 }
 
 template <typename T>
 int AudioEncoderDecoderIsacT<T>::IncomingPacket(const uint8_t* payload,
                                                 size_t payload_len,
                                                 uint16_t rtp_sequence_number,
                                                 uint32_t rtp_timestamp,
                                                 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);
 }
 
 template <typename T>
 int AudioEncoderDecoderIsacT<T>::ErrorCode() {
   CriticalSectionScoped cs(state_lock_.get());
   return T::GetErrorCode(isac_state_);
 }
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_AUDIO_ENCODER_ISAC_T_IMPL_H_