Convert channel counts to size_t.

webrtc/modules/audio_coding/neteq/neteq_impl.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 133 matching lines @@
       InsertPacketInternal(rtp_header, kSyncPayload, receive_timestamp, true);
 
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
   return kOK;
 }
 
 int NetEqImpl::GetAudio(size_t max_length, int16_t* output_audio,
-                        size_t* samples_per_channel, int* num_channels,
+                        size_t* samples_per_channel, size_t* num_channels,
                         NetEqOutputType* type) {
   CriticalSectionScoped lock(crit_sect_.get());
   int error = GetAudioInternal(max_length, output_audio, samples_per_channel,
                                num_channels);
   if (error != 0) {
     error_code_ = error;
     return kFail;
   }
   if (type) {
     *type = LastOutputType();
@@ skipping 571 matching lines @@
     // but don't update statistics.
     delay_manager_->set_last_pack_cng_or_dtmf(0);
     delay_manager_->ResetPacketIatCount();
   }
   return 0;
 }
 
 int NetEqImpl::GetAudioInternal(size_t max_length,
                                 int16_t* output,
                                 size_t* samples_per_channel,
-                                int* num_channels) {
+                                size_t* num_channels) {
   PacketList packet_list;
   DtmfEvent dtmf_event;
   Operations operation;
   bool play_dtmf;
   int return_value = GetDecision(&operation, &packet_list, &dtmf_event,
                                  &play_dtmf);
   if (return_value != 0) {
     last_mode_ = kModeError;
     return return_value;
   }
@@ skipping 103 matching lines @@
   size_t num_output_samples = output_size_samples_ * sync_buffer_->Channels();
   if (num_output_samples > max_length) {
     LOG(LS_WARNING) << "Output array is too short. " << max_length << " < " <<
         output_size_samples_ << " * " << sync_buffer_->Channels();
     num_output_samples = max_length;
     num_output_samples_per_channel = max_length / sync_buffer_->Channels();
   }
   const size_t samples_from_sync =
       sync_buffer_->GetNextAudioInterleaved(num_output_samples_per_channel,
                                             output);
-  *num_channels = static_cast<int>(sync_buffer_->Channels());
+  *num_channels = sync_buffer_->Channels();
   if (sync_buffer_->FutureLength() < expand_->overlap_length()) {
     // The sync buffer should always contain |overlap_length| samples, but now
     // too many samples have been extracted. Reinstall the |overlap_length|
     // lookahead by moving the index.
     const size_t missing_lookahead_samples =
         expand_->overlap_length() - sync_buffer_->FutureLength();
     RTC_DCHECK_GE(sync_buffer_->next_index(), missing_lookahead_samples);
     sync_buffer_->set_next_index(sync_buffer_->next_index() -
                                  missing_lookahead_samples);
   }
@@ skipping 1163 matching lines @@
 
 void NetEqImpl::CreateDecisionLogic() {
   decision_logic_.reset(DecisionLogic::Create(fs_hz_, output_size_samples_,
                                               playout_mode_,
                                               decoder_database_.get(),
                                               *packet_buffer_.get(),
                                               delay_manager_.get(),
                                               buffer_level_filter_.get()));
 }
 }  // namespace webrtc