Convert channel counts to size_t.

webrtc/modules/audio_processing/gain_control_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 57 matching lines @@
 
 int GainControlImpl::ProcessRenderAudio(AudioBuffer* audio) {
   rtc::CritScope cs(crit_render_);
   if (!is_component_enabled()) {
     return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
 
   render_queue_buffer_.resize(0);
-  for (int i = 0; i < num_handles(); i++) {
+  for (size_t i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int err =
         WebRtcAgc_GetAddFarendError(my_handle, audio->num_frames_per_band());
 
     if (err != AudioProcessing::kNoError)
       return GetHandleError(my_handle);
 
     // Buffer the samples in the render queue.
     render_queue_buffer_.insert(
         render_queue_buffer_.end(), audio->mixed_low_pass_data(),
@@ skipping 18 matching lines @@
   rtc::CritScope cs(crit_capture_);
 
   if (!is_component_enabled()) {
     return;
   }
 
   while (render_signal_queue_->Remove(&capture_queue_buffer_)) {
     size_t buffer_index = 0;
     const size_t num_frames_per_band =
         capture_queue_buffer_.size() / num_handles();
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       WebRtcAgc_AddFarend(my_handle, &capture_queue_buffer_[buffer_index],
                           num_frames_per_band);
 
       buffer_index += num_frames_per_band;
     }
   }
 }
 
 int GainControlImpl::AnalyzeCaptureAudio(AudioBuffer* audio) {
   rtc::CritScope cs(crit_capture_);
 
   if (!is_component_enabled()) {
     return AudioProcessing::kNoError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
   int err = AudioProcessing::kNoError;
 
   if (mode_ == kAdaptiveAnalog) {
     capture_levels_.assign(num_handles(), analog_capture_level_);
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       err = WebRtcAgc_AddMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
           audio->num_frames_per_band());
 
       if (err != AudioProcessing::kNoError) {
         return GetHandleError(my_handle);
       }
     }
   } else if (mode_ == kAdaptiveDigital) {
 
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       Handle* my_handle = static_cast<Handle*>(handle(i));
       int32_t capture_level_out = 0;
 
       err = WebRtcAgc_VirtualMic(
           my_handle,
           audio->split_bands(i),
           audio->num_bands(),
           audio->num_frames_per_band(),
           analog_capture_level_,
           &capture_level_out);
@@ skipping 18 matching lines @@
   }
 
   if (mode_ == kAdaptiveAnalog && !was_analog_level_set_) {
     return AudioProcessing::kStreamParameterNotSetError;
   }
 
   assert(audio->num_frames_per_band() <= 160);
   assert(audio->num_channels() == num_handles());
 
   stream_is_saturated_ = false;
-  for (int i = 0; i < num_handles(); i++) {
+  for (size_t i = 0; i < num_handles(); i++) {
     Handle* my_handle = static_cast<Handle*>(handle(i));
     int32_t capture_level_out = 0;
     uint8_t saturation_warning = 0;
 
     // The call to stream_has_echo() is ok from a deadlock perspective
     // as the capture lock is allready held.
     int err = WebRtcAgc_Process(
         my_handle,
         audio->split_bands_const(i),
         audio->num_bands(),
@@ skipping 10 matching lines @@
 
     capture_levels_[i] = capture_level_out;
     if (saturation_warning == 1) {
       stream_is_saturated_ = true;
     }
   }
 
   if (mode_ == kAdaptiveAnalog) {
     // Take the analog level to be the average across the handles.
     analog_capture_level_ = 0;
-    for (int i = 0; i < num_handles(); i++) {
+    for (size_t i = 0; i < num_handles(); i++) {
       analog_capture_level_ += capture_levels_[i];
     }
 
     analog_capture_level_ /= num_handles();
   }
 
   was_analog_level_set_ = false;
   return AudioProcessing::kNoError;
 }
 
@@ skipping 190 matching lines @@
   //assert(target_level_dbfs_ <= 0);
   //config.targetLevelDbfs = static_cast<int16_t>(-target_level_dbfs_);
   config.targetLevelDbfs = static_cast<int16_t>(target_level_dbfs_);
   config.compressionGaindB =
       static_cast<int16_t>(compression_gain_db_);
   config.limiterEnable = limiter_enabled_;
 
   return WebRtcAgc_set_config(static_cast<Handle*>(handle), config);
 }
 
-int GainControlImpl::num_handles_required() const {
+size_t GainControlImpl::num_handles_required() const {
   // Not locked as it only relies on APM public API which is threadsafe.
   return apm_->num_proc_channels();
 }
 
 int GainControlImpl::GetHandleError(void* handle) const {
   // The AGC has no get_error() function.
   // (Despite listing errors in its interface...)
   assert(handle != NULL);
   return AudioProcessing::kUnspecifiedError;
 }
 }  // namespace webrtc