[NOT FOR REVIEW] Convert channel counts to size_t.

webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc

Index: webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
index 9eb7a11524d4b13fcd1a23c82c9e0387f35fa99f..8f712f67626148dc1314756d0efef5de329be1e8 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -49,7 +49,7 @@ AudioEncoderG722::AudioEncoderG722(const Config& config)
   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);
   }
@@ -67,7 +67,7 @@ int AudioEncoderG722::RtpTimestampRateHz() const {
   return kSampleRateHz / 2;
 }
 
-int AudioEncoderG722::NumChannels() const {
+size_t AudioEncoderG722::NumChannels() const {
   return num_channels_;
 }
 
@@ -101,7 +101,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   // 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.
@@ -113,7 +113,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   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>());
@@ -124,12 +124,12 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   // 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];
   }