Match existing type usage better.

webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.cc

 /*
  *  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 97 matching lines @@
 }
 
 int AudioEncoderOpus::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderOpus::MaxEncodedBytes() const {
   // Calculate the number of bytes we expect the encoder to produce,
   // then multiply by two to give a wide margin for error.
   int frame_size_ms = num_10ms_frames_per_packet_ * 10;
-  int bytes_per_millisecond = bitrate_bps_ / (1000 * 8) + 1;
-  size_t approx_encoded_bytes =
-      static_cast<size_t>(frame_size_ms * bytes_per_millisecond);
+  size_t bytes_per_millisecond =
+      static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
+  size_t approx_encoded_bytes = frame_size_ms * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }
 
 int AudioEncoderOpus::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderOpus::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
@@ skipping 68 matching lines @@
            static_cast<size_t>(num_10ms_frames_per_packet_) *
            samples_per_10ms_frame_);
   int16_t status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
       rtc::CheckedDivExact(CastInt16(input_buffer_.size()),
                            static_cast<int16_t>(num_channels_)),
       ClampInt16(max_encoded_bytes), encoded);
   CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
-  info.encoded_bytes = status;
+  info.encoded_bytes = static_cast<size_t>(status);
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (status > 0);
   return info;
 }
 
 namespace {
 AudioEncoderOpus::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderOpus::Config config;
@@ skipping 36 matching lines @@
   return Reconstruct(conf);
 }
 
 bool AudioEncoderMutableOpus::SetMaxPlaybackRate(int frequency_hz) {
   auto conf = config();
   conf.max_playback_rate_hz = frequency_hz;
   return Reconstruct(conf);
 }
 
 }  // namespace webrtc