[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.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.
  */
 
 #include "webrtc/modules/audio_coding/codecs/g711/include/audio_encoder_pcm.h"
 
 #include <limits>
 
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/g711/include/g711_interface.h"
 
 namespace webrtc {
 
 namespace {
-int16_t NumSamplesPerFrame(int num_channels,
+int16_t NumSamplesPerFrame(size_t num_channels,
                            int frame_size_ms,
                            int sample_rate_hz) {
-  int samples_per_frame = num_channels * frame_size_ms * sample_rate_hz / 1000;
-  CHECK_LE(samples_per_frame, std::numeric_limits<int16_t>::max())
+  size_t samples_per_frame =
+      num_channels * frame_size_ms * sample_rate_hz / 1000;
+  CHECK_LE(samples_per_frame,
+           static_cast<size_t>(std::numeric_limits<int16_t>::max()))
       << "Frame size too large.";
   return static_cast<int16_t>(samples_per_frame);
 }
 }  // namespace
 
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
 
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
@@ skipping 12 matching lines @@
   speech_buffer_.reserve(full_frame_samples_);
 }
 
 AudioEncoderPcm::~AudioEncoderPcm() {
 }
 
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
-int AudioEncoderPcm::NumChannels() const {
+size_t AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderPcm::MaxEncodedBytes() const {
   return full_frame_samples_ * BytesPerSample();
 }
 
 size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
@@ skipping 67 matching lines @@
     : AudioEncoderMutableImpl<AudioEncoderPcmU>(
           CreateConfig<AudioEncoderPcmU>(codec_inst)) {
 }
 
 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderPcmA>(
           CreateConfig<AudioEncoderPcmA>(codec_inst)) {
 }
 
 }  // namespace webrtc