Update audio code to use size_t more correctly,

webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.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/ilbc/interface/audio_encoder_ilbc.h"
 
 #include <cstring>
 #include <limits>
 #include "webrtc/base/checks.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/interface/ilbc.h"
 
 namespace webrtc {
 
 namespace {
 
 const int kSampleRateHz = 8000;
 
 }  // namespace
 
+// static
+const size_t AudioEncoderIlbc::kMaxSamplesPerPacket;
+
 bool AudioEncoderIlbc::Config::IsOk() const {
   return (frame_size_ms == 20 || frame_size_ms == 30 || frame_size_ms == 40 ||
           frame_size_ms == 60) &&
-      (kSampleRateHz / 100 * (frame_size_ms / 10)) <= kMaxSamplesPerPacket;
+      static_cast<size_t>(kSampleRateHz / 100 * (frame_size_ms / 10)) <=
+          kMaxSamplesPerPacket;
 }
 
 AudioEncoderIlbc::AudioEncoderIlbc(const Config& config)
     : 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) {
   CHECK(config.IsOk());
   CHECK_EQ(0, WebRtcIlbcfix_EncoderCreate(&encoder_));
   const int encoder_frame_size_ms = config.frame_size_ms > 30
                                         ? config.frame_size_ms / 2
                                         : config.frame_size_ms;
   CHECK_EQ(0, WebRtcIlbcfix_EncoderInit(encoder_, encoder_frame_size_ms));
 }
 
 AudioEncoderIlbc::~AudioEncoderIlbc() {
   CHECK_EQ(0, WebRtcIlbcfix_EncoderFree(encoder_));
 }
 
 int AudioEncoderIlbc::SampleRateHz() const {
   return kSampleRateHz;
 }
 
 int AudioEncoderIlbc::NumChannels() const {
   return 1;
 }
 
 size_t AudioEncoderIlbc::MaxEncodedBytes() const {
   return RequiredOutputSizeBytes();
 }
 
-int AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderIlbc::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderIlbc::Max10MsFramesInAPacket() const {
+size_t AudioEncoderIlbc::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderIlbc::GetTargetBitrate() const {
   switch (num_10ms_frames_per_packet_) {
     case 2: case 4:
       // 38 bytes per frame of 20 ms => 15200 bits/s.
       return 15200;
     case 3: case 6:
       // 50 bytes per frame of 30 ms => (approx) 13333 bits/s.
@@ skipping 28 matching lines @@
   // Encode buffered input.
   DCHECK_EQ(num_10ms_frames_buffered_, num_10ms_frames_per_packet_);
   num_10ms_frames_buffered_ = 0;
   const int output_len = WebRtcIlbcfix_Encode(
       encoder_,
       input_buffer_,
       kSampleRateHz / 100 * num_10ms_frames_per_packet_,
       encoded);
   CHECK_GE(output_len, 0);
   EncodedInfo info;
-  info.encoded_bytes = output_len;
+  info.encoded_bytes = static_cast<size_t>(output_len);
   DCHECK_EQ(info.encoded_bytes, RequiredOutputSizeBytes());
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
   return info;
 }
 
 size_t AudioEncoderIlbc::RequiredOutputSizeBytes() const {
   switch (num_10ms_frames_per_packet_) {
     case 2:   return 38;
     case 3:   return 50;
@@ skipping 10 matching lines @@
   config.payload_type = codec_inst.pltype;
   return config;
 }
 }  // namespace
 
 AudioEncoderMutableIlbc::AudioEncoderMutableIlbc(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderIlbc>(CreateConfig(codec_inst)) {
 }
 
 }  // namespace webrtc