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.
  */
 
@@ skipping 16 matching lines @@
   RemixAndResample(src_frame.data_, src_frame.samples_per_channel_,
                    src_frame.num_channels_, src_frame.sample_rate_hz_,
                    resampler, 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 RemixAndResample(const int16_t* src_data,
                       size_t samples_per_channel,
-                      int num_channels,
+                      size_t num_channels,
                       int sample_rate_hz,
                       PushResampler<int16_t>* resampler,
                       AudioFrame* dst_frame) {
   const int16_t* audio_ptr = src_data;
-  int audio_ptr_num_channels = num_channels;
+  size_t audio_ptr_num_channels = num_channels;
   int16_t mono_audio[AudioFrame::kMaxDataSizeSamples];
 
   // Downmix before resampling.
   if (num_channels == 2 && dst_frame->num_channels_ == 1) {
     AudioFrameOperations::StereoToMono(src_data, samples_per_channel,
                                        mono_audio);
     audio_ptr = mono_audio;
     audio_ptr_num_channels = 1;
   }
 
   if (resampler->InitializeIfNeeded(sample_rate_hz, dst_frame->sample_rate_hz_,
                                     audio_ptr_num_channels) == -1) {
     LOG(LS_ERROR) << "InitializeIfNeeded failed: sample_rate_hz = "
                   << sample_rate_hz << ", dst_frame->sample_rate_hz_ = "
                   << dst_frame->sample_rate_hz_
                   << ", audio_ptr_num_channels = " << audio_ptr_num_channels;
     assert(false);
   }
 
   const size_t src_length = 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(LS_ERROR) << "Resample failed: audio_ptr = " << audio_ptr
                   << ", src_length = " << src_length
                   << ", dst_frame->data_ = " << 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 (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);
   }
 }
 
 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