[NOT FOR REVIEW] Convert channel counts to size_t.

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 49 matching lines @@
   return true;
 }
 
 AudioEncoderOpus::AudioEncoderOpus(const Config& config)
     : num_10ms_frames_per_packet_(
           static_cast<size_t>(rtc::CheckedDivExact(config.frame_size_ms, 10))),
       num_channels_(config.num_channels),
       payload_type_(config.payload_type),
       application_(config.application),
       dtx_enabled_(config.dtx_enabled),
-      samples_per_10ms_frame_(static_cast<size_t>(
-          rtc::CheckedDivExact(kSampleRateHz, 100) * num_channels_)),
+      samples_per_10ms_frame_(
+          rtc::CheckedDivExact(kSampleRateHz, 100) * num_channels_),
       packet_loss_rate_(0.0) {
   CHECK(config.IsOk());
   input_buffer_.reserve(num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
   CHECK_EQ(0, WebRtcOpus_EncoderCreate(&inst_, num_channels_, application_));
   SetTargetBitrate(config.bitrate_bps);
   if (config.fec_enabled) {
     CHECK_EQ(0, WebRtcOpus_EnableFec(inst_));
   } else {
     CHECK_EQ(0, WebRtcOpus_DisableFec(inst_));
   }
   CHECK_EQ(0,
            WebRtcOpus_SetMaxPlaybackRate(inst_, config.max_playback_rate_hz));
   CHECK_EQ(0, WebRtcOpus_SetComplexity(inst_, config.complexity));
   if (config.dtx_enabled) {
     CHECK_EQ(0, WebRtcOpus_EnableDtx(inst_));
   } else {
     CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
 }
 
 AudioEncoderOpus::~AudioEncoderOpus() {
   CHECK_EQ(0, WebRtcOpus_EncoderFree(inst_));
 }
 
 int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-int AudioEncoderOpus::NumChannels() const {
+size_t 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.
   size_t bytes_per_millisecond =
       static_cast<size_t>(bitrate_bps_ / (1000 * 8) + 1);
   size_t approx_encoded_bytes =
       num_10ms_frames_per_packet_ * 10 * bytes_per_millisecond;
@@ skipping 75 matching lines @@
   input_buffer_.insert(input_buffer_.end(), audio,
                        audio + samples_per_10ms_frame_);
   if (input_buffer_.size() <
       (num_10ms_frames_per_packet_ * samples_per_10ms_frame_)) {
     return EncodedInfo();
   }
   CHECK_EQ(input_buffer_.size(),
            num_10ms_frames_per_packet_ * samples_per_10ms_frame_);
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
-      rtc::CheckedDivExact(input_buffer_.size(),
-                           static_cast<size_t>(num_channels_)),
+      rtc::CheckedDivExact(input_buffer_.size(), num_channels_),
       max_encoded_bytes, encoded);
   CHECK_GE(status, 0);  // Fails only if fed invalid data.
   input_buffer_.clear();
   EncodedInfo info;
   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;
@@ skipping 41 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