[NOT FOR REVIEW] Convert channel counts to size_t.

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.
  */
 
 #include "webrtc/voice_engine/utility.h"
 
 #include "webrtc/common_audio/resampler/include/push_resampler.h"
 #include "webrtc/common_audio/signal_processing/include/signal_processing_library.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/interface/module_common_types.h"
 #include "webrtc/modules/utility/interface/audio_frame_operations.h"
 #include "webrtc/system_wrappers/interface/logging.h"
 #include "webrtc/voice_engine/voice_engine_defines.h"
 
 namespace webrtc {
 namespace voe {
 
 // TODO(ajm): There is significant overlap between RemixAndResample and
 // ConvertToCodecFormat. Consolidate using AudioConverter.
 void RemixAndResample(const AudioFrame& src_frame,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
   const int16_t* audio_ptr = src_frame.data_;
-  int audio_ptr_num_channels = src_frame.num_channels_;
+  size_t audio_ptr_num_channels = src_frame.num_channels_;
   int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
 
   // Downmix before resampling.
   if (src_frame.num_channels_ == 2 && dst_frame->num_channels_ == 1) {
     AudioFrameOperations::StereoToMono(src_frame.data_,
                                        src_frame.samples_per_channel_,
                                        mono_audio);
     audio_ptr = mono_audio;
     audio_ptr_num_channels = 1;
   }
 
   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 size_t src_length = src_frame.samples_per_channel_ *
-                         audio_ptr_num_channels;
+  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_ =
-      static_cast<size_t>(out_length / audio_ptr_num_channels);
+  dst_frame->samples_per_channel_ = 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,
                               size_t samples_per_channel,
-                              int num_channels,
+                              size_t num_channels,
                               int sample_rate_hz,
-                              int codec_num_channels,
+                              size_t 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();
 
   // Never upsample the capture signal here. This should be done at the
@@ skipping 20 matching lines @@
   }
 
   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_ = static_cast<size_t>(out_length / num_channels);
+  dst_af->samples_per_channel_ = 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,
+                size_t target_channel,
                 const int16_t source[],
-                int source_channel,
+                size_t source_channel,
                 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 (size_t i = 0; i < source_len; ++i) {
       left = source[i] + target[i * 2];
@@ skipping 12 matching lines @@
     int32_t temp = 0;
     for (size_t i = 0; i < source_len; ++i) {
       temp = source[i] + target[i];
       target[i] = WebRtcSpl_SatW32ToW16(temp);
     }
   }
 }
 
 }  // namespace voe
 }  // namespace webrtc