[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.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 31 matching lines @@
       payload_type_(config.payload_type),
       num_10ms_frames_per_packet_(
           static_cast<size_t>(config.frame_size_ms / 10)),
       num_10ms_frames_buffered_(0),
       first_timestamp_in_buffer_(0),
       encoders_(new EncoderState[num_channels_]),
       interleave_buffer_(2 * num_channels_) {
   CHECK(config.IsOk());
   const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
     encoders_[i].encoded_buffer.SetSize(samples_per_channel / 2);
   }
 }
 
 AudioEncoderG722::~AudioEncoderG722() {}
 
 int AudioEncoderG722::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 int AudioEncoderG722::RtpTimestampRateHz() const {
   // The RTP timestamp rate for G.722 is 8000 Hz, even though it is a 16 kHz
   // codec.
   return kSampleRateHz / 2;
 }
 
-int AudioEncoderG722::NumChannels() const {
+size_t AudioEncoderG722::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderG722::MaxEncodedBytes() const {
   return SamplesPerChannel() / 2 * num_channels_;
 }
 
 size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
@@ skipping 13 matching lines @@
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   CHECK_GE(max_encoded_bytes, MaxEncodedBytes());
 
   if (num_10ms_frames_buffered_ == 0)
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Deinterleave samples and save them in each channel's buffer.
   const size_t start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
   for (size_t i = 0; i < kSampleRateHz / 100; ++i)
-    for (int j = 0; j < num_channels_; ++j)
+    for (size_t j = 0; j < num_channels_; ++j)
       encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
 
   // If we don't yet have enough samples for a packet, we're done for now.
   if (++num_10ms_frames_buffered_ < num_10ms_frames_per_packet_) {
     return EncodedInfo();
   }
 
   // Encode each channel separately.
   CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
   const size_t samples_per_channel = SamplesPerChannel();
-  for (int i = 0; i < num_channels_; ++i) {
+  for (size_t i = 0; i < num_channels_; ++i) {
     const size_t encoded = WebRtcG722_Encode(
         encoders_[i].encoder, encoders_[i].speech_buffer.get(),
         samples_per_channel, encoders_[i].encoded_buffer.data<uint8_t>());
     CHECK_EQ(encoded, samples_per_channel / 2);
   }
 
   // Interleave the encoded bytes of the different channels. Each separate
   // channel and the interleaved stream encodes two samples per byte, most
   // significant half first.
   for (size_t i = 0; i < samples_per_channel / 2; ++i) {
-    for (int j = 0; j < num_channels_; ++j) {
+    for (size_t j = 0; j < num_channels_; ++j) {
       uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
       interleave_buffer_.data()[j] = two_samples >> 4;
       interleave_buffer_.data()[num_channels_ + j] = two_samples & 0xf;
     }
-    for (int j = 0; j < num_channels_; ++j)
+    for (size_t j = 0; j < num_channels_; ++j)
       encoded[i * num_channels_ + j] = interleave_buffer_.data()[2 * j] << 4 |
                                        interleave_buffer_.data()[2 * j + 1];
   }
   EncodedInfo info;
   info.encoded_bytes = samples_per_channel / 2 * num_channels_;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   return info;
 }
 
 size_t AudioEncoderG722::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }
 
 namespace {
 AudioEncoderG722::Config CreateConfig(const CodecInst& codec_inst) {
   AudioEncoderG722::Config config;
   config.num_channels = codec_inst.channels;
   config.frame_size_ms = codec_inst.pacsize / 16;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 }  // namespace
 
 AudioEncoderMutableG722::AudioEncoderMutableG722(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderG722>(CreateConfig(codec_inst)) {
 }
 
 }  // namespace webrtc