Update audio code to use size_t more correctly,

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 a0d1720e460f9c90231ca583f9d001df6e8cc826..9eb7a11524d4b13fcd1a23c82c9e0387f35fa99f 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.cc
@@ -19,7 +19,7 @@ namespace webrtc {
 
 namespace {
 
-const int kSampleRateHz = 16000;
+const size_t kSampleRateHz = 16000;
 
 }  // namespace
 
@@ -40,13 +40,14 @@ AudioEncoderG722::EncoderState::~EncoderState() {
 AudioEncoderG722::AudioEncoderG722(const Config& config)
     : num_channels_(config.num_channels),
       payload_type_(config.payload_type),
-      num_10ms_frames_per_packet_(config.frame_size_ms / 10),
+      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 int samples_per_channel =
+  const size_t samples_per_channel =
       kSampleRateHz / 100 * num_10ms_frames_per_packet_;
   for (int i = 0; i < num_channels_; ++i) {
     encoders_[i].speech_buffer.reset(new int16_t[samples_per_channel]);
@@ -71,14 +72,14 @@ int AudioEncoderG722::NumChannels() const {
 }
 
 size_t AudioEncoderG722::MaxEncodedBytes() const {
-  return static_cast<size_t>(SamplesPerChannel() / 2 * num_channels_);
+  return SamplesPerChannel() / 2 * num_channels_;
 }
 
-int AudioEncoderG722::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderG722::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderG722::Max10MsFramesInAPacket() const {
+size_t AudioEncoderG722::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
@@ -98,8 +99,8 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
     first_timestamp_in_buffer_ = rtp_timestamp;
 
   // Deinterleave samples and save them in each channel's buffer.
-  const int start = kSampleRateHz / 100 * num_10ms_frames_buffered_;
-  for (int i = 0; i < kSampleRateHz / 100; ++i)
+  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)
       encoders_[j].speech_buffer[start + i] = audio[i * num_channels_ + j];
 
@@ -111,19 +112,18 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   // Encode each channel separately.
   CHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
-  const int samples_per_channel = SamplesPerChannel();
+  const size_t samples_per_channel = SamplesPerChannel();
   for (int i = 0; i < num_channels_; ++i) {
-    const int encoded = WebRtcG722_Encode(
+    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_GE(encoded, 0);
     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 (int i = 0; i < samples_per_channel / 2; ++i) {
+  for (size_t i = 0; i < samples_per_channel / 2; ++i) {
     for (int j = 0; j < num_channels_; ++j) {
       uint8_t two_samples = encoders_[j].encoded_buffer.data()[i];
       interleave_buffer_.data()[j] = two_samples >> 4;
@@ -140,7 +140,7 @@ AudioEncoder::EncodedInfo AudioEncoderG722::EncodeInternal(
   return info;
 }
 
-int AudioEncoderG722::SamplesPerChannel() const {
+size_t AudioEncoderG722::SamplesPerChannel() const {
   return kSampleRateHz / 100 * num_10ms_frames_per_packet_;
 }