Convert channel counts to size_t.

webrtc/modules/audio_processing/echo_cancellation_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 81 matching lines @@
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_reverse_channels());
 
   int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   render_queue_buffer_.clear();
-  for (int i = 0; i < apm_->num_output_channels(); i++) {
-    for (int j = 0; j < audio->num_channels(); j++) {
+  for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+    for (size_t j = 0; j < audio->num_channels(); j++) {
       Handle* my_handle = static_cast<Handle*>(handle(handle_index));
       // Retrieve any error code produced by the buffering of the farend
       // signal
       err = WebRtcAec_GetBufferFarendError(
           my_handle, audio->split_bands_const_f(j)[kBand0To8kHz],
           audio->num_frames_per_band());
 
       if (err != AudioProcessing::kNoError) {
         return MapError(err);  // TODO(ajm): warning possible?
       }
@@ skipping 25 matching lines @@
   if (!is_component_enabled()) {
     return;
   }
 
   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
     size_t handle_index = 0;
     size_t buffer_index = 0;
     const size_t num_frames_per_band =
         capture_queue_buffer_.size() /
         (apm_->num_output_channels() * apm_->num_reverse_channels());
-    for (int i = 0; i < apm_->num_output_channels(); i++) {
-      for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+    for (size_t i = 0; i < apm_->num_output_channels(); i++) {
+      for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
         Handle* my_handle = static_cast<Handle*>(handle(handle_index));
         WebRtcAec_BufferFarend(my_handle, &capture_queue_buffer_[buffer_index],
                                num_frames_per_band);
 
         buffer_index += num_frames_per_band;
         handle_index++;
       }
     }
   }
 }
@@ skipping 13 matching lines @@
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == apm_->num_proc_channels());
 
   int err = AudioProcessing::kNoError;
 
   // The ordering convention must be followed to pass to the correct AEC.
   size_t handle_index = 0;
   stream_has_echo_ = false;
-  for (int i = 0; i < audio->num_channels(); i++) {
-    for (int j = 0; j < apm_->num_reverse_channels(); j++) {
+  for (size_t i = 0; i < audio->num_channels(); i++) {
+    for (size_t j = 0; j < apm_->num_reverse_channels(); j++) {
       Handle* my_handle = handle(handle_index);
       err = WebRtcAec_Process(my_handle, audio->split_bands_const_f(i),
                               audio->num_bands(), audio->split_bands_f(i),
                               audio->num_frames_per_band(),
                               apm_->stream_delay_ms(), stream_drift_samples_);
 
       if (err != AudioProcessing::kNoError) {
         err = MapError(err);
         // TODO(ajm): Figure out how to return warnings properly.
         if (err != AudioProcessing::kBadStreamParameterWarning) {
@@ skipping 286 matching lines @@
   config.delay_logging = delay_logging_enabled_;
   WebRtcAec_enable_extended_filter(
       WebRtcAec_aec_core(static_cast<Handle*>(handle)),
       extended_filter_enabled_ ? 1 : 0);
   WebRtcAec_enable_delay_agnostic(
       WebRtcAec_aec_core(static_cast<Handle*>(handle)),
       delay_agnostic_enabled_ ? 1 : 0);
   return WebRtcAec_set_config(static_cast<Handle*>(handle), config);
 }
 
-int EchoCancellationImpl::num_handles_required() const {
+size_t EchoCancellationImpl::num_handles_required() const {
   // Not locked as it only relies on APM public API which is threadsafe.
   return apm_->num_output_channels() * apm_->num_reverse_channels();
 }
 
 int EchoCancellationImpl::GetHandleError(void* handle) const {
   // Not locked as it does not rely on anything in the state.
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc