Update a ton of audio code to use size_t more correctly and in general reduce

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.
  */
 
@@ skipping 19 matching lines @@
 }  // namespace
 
 bool AudioEncoderPcm::Config::IsOk() const {
   return (frame_size_ms % 10 == 0) && (num_channels >= 1);
 }
 
 AudioEncoderPcm::AudioEncoderPcm(const Config& config, int sample_rate_hz)
     : sample_rate_hz_(sample_rate_hz),
       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)),
       full_frame_samples_(NumSamplesPerFrame(config.num_channels,
                                              config.frame_size_ms,
                                              sample_rate_hz_)),
       first_timestamp_in_buffer_(0) {
   CHECK_GT(sample_rate_hz, 0) << "Sample rate must be larger than 0 Hz";
   CHECK_EQ(config.frame_size_ms % 10, 0)
       << "Frame size must be an integer multiple of 10 ms.";
   speech_buffer_.reserve(full_frame_samples_);
 }
 
 AudioEncoderPcm::~AudioEncoderPcm() {
 }
 
 int AudioEncoderPcm::SampleRateHz() const {
   return sample_rate_hz_;
 }
 
 int AudioEncoderPcm::NumChannels() const {
   return num_channels_;
 }
 
 size_t AudioEncoderPcm::MaxEncodedBytes() const {
   return full_frame_samples_ * BytesPerSample();
 }
 
-int AudioEncoderPcm::Num10MsFramesInNextPacket() const {
+size_t AudioEncoderPcm::Num10MsFramesInNextPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
-int AudioEncoderPcm::Max10MsFramesInAPacket() const {
+size_t AudioEncoderPcm::Max10MsFramesInAPacket() const {
   return num_10ms_frames_per_packet_;
 }
 
 int AudioEncoderPcm::GetTargetBitrate() const {
   return 8 * BytesPerSample() * SampleRateHz() * NumChannels();
 }
 
 AudioEncoder::EncodedInfo AudioEncoderPcm::EncodeInternal(
     uint32_t rtp_timestamp,
     const int16_t* audio,
     size_t max_encoded_bytes,
     uint8_t* encoded) {
   const int num_samples = SampleRateHz() / 100 * NumChannels();
   if (speech_buffer_.empty()) {
     first_timestamp_in_buffer_ = rtp_timestamp;
   }
   for (int i = 0; i < num_samples; ++i) {
     speech_buffer_.push_back(audio[i]);
   }
   if (speech_buffer_.size() < full_frame_samples_) {
     return EncodedInfo();
   }
   CHECK_EQ(speech_buffer_.size(), full_frame_samples_);
   CHECK_GE(max_encoded_bytes, full_frame_samples_);
   EncodedInfo info;
   info.encoded_timestamp = first_timestamp_in_buffer_;
   info.payload_type = payload_type_;
-  int16_t ret = EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
-  CHECK_GE(ret, 0);
-  info.encoded_bytes = static_cast<size_t>(ret);
+  info.encoded_bytes =
+      EncodeCall(&speech_buffer_[0], full_frame_samples_, encoded);
   speech_buffer_.clear();
   return info;
 }
 
-int16_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
-                                     size_t input_len,
-                                     uint8_t* encoded) {
-  return WebRtcG711_EncodeA(audio, static_cast<int16_t>(input_len), encoded);
+size_t AudioEncoderPcmA::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeA(audio, input_len, encoded);
 }
 
 int AudioEncoderPcmA::BytesPerSample() const {
   return 1;
 }
 
-int16_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
-                                     size_t input_len,
-                                     uint8_t* encoded) {
-  return WebRtcG711_EncodeU(audio, static_cast<int16_t>(input_len), encoded);
+size_t AudioEncoderPcmU::EncodeCall(const int16_t* audio,
+                                    size_t input_len,
+                                    uint8_t* encoded) {
+  return WebRtcG711_EncodeU(audio, input_len, encoded);
 }
 
 int AudioEncoderPcmU::BytesPerSample() const {
   return 1;
 }
 
 namespace {
 template <typename T>
 typename T::Config CreateConfig(const CodecInst& codec_inst) {
   typename T::Config config;
   config.frame_size_ms = codec_inst.pacsize / 8;
   config.num_channels = codec_inst.channels;
   config.payload_type = codec_inst.pltype;
   return config;
 }
 }  // namespace
 
 AudioEncoderMutablePcmU::AudioEncoderMutablePcmU(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderPcmU>(
           CreateConfig<AudioEncoderPcmU>(codec_inst)) {
 }
 
 AudioEncoderMutablePcmA::AudioEncoderMutablePcmA(const CodecInst& codec_inst)
     : AudioEncoderMutableImpl<AudioEncoderPcmA>(
           CreateConfig<AudioEncoderPcmA>(codec_inst)) {
 }
 
 }  // namespace webrtc