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 96 matching lines @@
       static_cast<size_t>(config_.bitrate_bps / (1000 * 8) + 1);
   const size_t approx_encoded_bytes =
       Num10msFramesPerPacket() * 10 * bytes_per_millisecond;
   return 2 * approx_encoded_bytes;
 }
 
 int AudioEncoderOpus::SampleRateHz() const {
   return kSampleRateHz;
 }
 
-int AudioEncoderOpus::NumChannels() const {
+size_t AudioEncoderOpus::NumChannels() const {
   return config_.num_channels;
 }
 
 size_t AudioEncoderOpus::Num10MsFramesInNextPacket() const {
   return Num10msFramesPerPacket();
 }
 
 size_t AudioEncoderOpus::Max10MsFramesInAPacket() const {
   return Num10msFramesPerPacket();
 }
@@ skipping 12 matching lines @@
   RTC_DCHECK_EQ(SamplesPer10msFrame(), audio.size());
   input_buffer_.insert(input_buffer_.end(), audio.cbegin(), audio.cend());
   if (input_buffer_.size() <
       (Num10msFramesPerPacket() * SamplesPer10msFrame())) {
     return EncodedInfo();
   }
   RTC_CHECK_EQ(input_buffer_.size(),
                Num10msFramesPerPacket() * SamplesPer10msFrame());
   int status = WebRtcOpus_Encode(
       inst_, &input_buffer_[0],
-      rtc::CheckedDivExact(input_buffer_.size(),
-                           static_cast<size_t>(config_.num_channels)),
+      rtc::CheckedDivExact(input_buffer_.size(), config_.num_channels),
       rtc::saturated_cast<int16_t>(max_encoded_bytes), encoded);
   RTC_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 = config_.payload_type;
   info.send_even_if_empty = true;  // Allows Opus to send empty packets.
   info.speech = (status > 0);
   return info;
@@ skipping 86 matching lines @@
     RTC_CHECK_EQ(0, WebRtcOpus_DisableDtx(inst_));
   }
   RTC_CHECK_EQ(0,
                WebRtcOpus_SetPacketLossRate(
                    inst_, static_cast<int32_t>(packet_loss_rate_ * 100 + .5)));
   config_ = config;
   return true;
 }
 
 }  // namespace webrtc