Update audio code to use size_t more correctly, webrtc/voice_engine/ portion.

webrtc/voice_engine/utility.cc

 /*
  *  Copyright (c) 2012 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 29 matching lines @@
   }
 
   if (resampler->InitializeIfNeeded(src_frame.sample_rate_hz_,
                                     dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
     LOG_FERR3(LS_ERROR, InitializeIfNeeded, src_frame.sample_rate_hz_,
               dst_frame->sample_rate_hz_, audio_ptr_num_channels);
     assert(false);
   }
 
-  const int src_length = src_frame.samples_per_channel_ *
+  const size_t src_length = src_frame.samples_per_channel_ *
                          audio_ptr_num_channels;
   int out_length = resampler->Resample(audio_ptr, src_length, dst_frame->data_,
                                        AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
     LOG_FERR3(LS_ERROR, Resample, audio_ptr, src_length, dst_frame->data_);
     assert(false);
   }
-  dst_frame->samples_per_channel_ = out_length / audio_ptr_num_channels;
+  dst_frame->samples_per_channel_ =
+      static_cast<size_t>(out_length / audio_ptr_num_channels);
 
   // Upmix after resampling.
   if (src_frame.num_channels_ == 1 && dst_frame->num_channels_ == 2) {
     // The audio in dst_frame really is mono at this point; MonoToStereo will
     // set this back to stereo.
     dst_frame->num_channels_ = 1;
     AudioFrameOperations::MonoToStereo(dst_frame);
   }
 
   dst_frame->timestamp_ = src_frame.timestamp_;
   dst_frame->elapsed_time_ms_ = src_frame.elapsed_time_ms_;
   dst_frame->ntp_time_ms_ = src_frame.ntp_time_ms_;
 }
 
 void DownConvertToCodecFormat(const int16_t* src_data,
-                              int samples_per_channel,
+                              size_t samples_per_channel,
                               int num_channels,

[pbos-webrtc, 2015/07/14 08:15:10]

size_t for num_channels?

                               int sample_rate_hz,
                               int codec_num_channels,
                               int codec_rate_hz,
                               int16_t* mono_buffer,
                               PushResampler<int16_t>* resampler,
                               AudioFrame* dst_af) {
   assert(samples_per_channel <= kMaxMonoDataSizeSamples);
   assert(num_channels == 1 || num_channels == 2);
   assert(codec_num_channels == 1 || codec_num_channels == 2);
   dst_af->Reset();
@@ skipping 14 matching lines @@
   if (resampler->InitializeIfNeeded(
           sample_rate_hz, destination_rate, num_channels) != 0) {
     LOG_FERR3(LS_ERROR,
               InitializeIfNeeded,
               sample_rate_hz,
               destination_rate,
               num_channels);
     assert(false);
   }
 
-  const int in_length = samples_per_channel * num_channels;
+  const size_t in_length = samples_per_channel * num_channels;
   int out_length = resampler->Resample(
       src_data, in_length, dst_af->data_, AudioFrame::kMaxDataSizeSamples);
   if (out_length == -1) {
     LOG_FERR3(LS_ERROR, Resample, src_data, in_length, dst_af->data_);
     assert(false);
   }
 
-  dst_af->samples_per_channel_ = out_length / num_channels;
+  dst_af->samples_per_channel_ = static_cast<size_t>(out_length / num_channels);
   dst_af->sample_rate_hz_ = destination_rate;
   dst_af->num_channels_ = num_channels;
 }
 
 void MixWithSat(int16_t target[],
                 int target_channel,
                 const int16_t source[],
                 int source_channel,
-                int source_len) {
+                size_t source_len) {
   assert(target_channel == 1 || target_channel == 2);
   assert(source_channel == 1 || source_channel == 2);
 
   if (target_channel == 2 && source_channel == 1) {
     // Convert source from mono to stereo.
     int32_t left = 0;
     int32_t right = 0;
-    for (int i = 0; i < source_len; ++i) {
+    for (size_t i = 0; i < source_len; ++i) {
       left = source[i] + target[i * 2];
       right = source[i] + target[i * 2 + 1];
       target[i * 2] = WebRtcSpl_SatW32ToW16(left);
       target[i * 2 + 1] = WebRtcSpl_SatW32ToW16(right);
     }
   } else if (target_channel == 1 && source_channel == 2) {
     // Convert source from stereo to mono.
     int32_t temp = 0;
-    for (int i = 0; i < source_len / 2; ++i) {
+    for (size_t i = 0; i < source_len / 2; ++i) {
       temp = ((source[i * 2] + source[i * 2 + 1]) >> 1) + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   } else {
     int32_t temp = 0;
-    for (int i = 0; i < source_len; ++i) {
+    for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   }
 }
 
 }  // namespace voe
 }  // namespace webrtc