Added a query function for whether submodules are enabled

webrtc/modules/audio_processing/audio_processing_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 1139 matching lines @@
 }
 
 bool AudioProcessingImpl::rev_synthesis_needed() const {
   return (is_rev_processed() &&
           formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz);
 }
 
 bool AudioProcessingImpl::rev_analysis_needed() const {
   return formats_.rev_proc_format.sample_rate_hz() == kSampleRate32kHz &&
          (is_rev_processed() ||
-          public_submodules_->echo_cancellation->is_enabled() ||
-          public_submodules_->echo_control_mobile->is_enabled() ||
-          public_submodules_->gain_control->is_enabled());
+          public_submodules_->echo_cancellation
+              ->is_enabled_render_side_query() ||
+          public_submodules_->echo_control_mobile
+              ->is_enabled_render_side_query() ||
+          public_submodules_->gain_control->is_enabled_render_side_query());
 }
 
 bool AudioProcessingImpl::render_check_rev_conversion_needed() const {
   return rev_conversion_needed();
 }
 
 bool AudioProcessingImpl::rev_conversion_needed() const {
   return (formats_.api_format.reverse_input_stream() !=
           formats_.api_format.reverse_output_stream());
 }
@@ skipping 274 matching lines @@
   debug_dump_.capture.event_msg->mutable_config()->CopyFrom(config);
 
   RETURN_ON_ERR(WriteMessageToDebugFile(debug_dump_.debug_file.get(),
                                         &debug_dump_.num_bytes_left_for_log_,
                                         &crit_debug_, &debug_dump_.capture));
   return kNoError;
 }
 #endif  // WEBRTC_AUDIOPROC_DEBUG_DUMP
 
 }  // namespace webrtc