Added functionality for specifying the initial signal level to use for the gain estimation in the l…

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 972 matching lines @@
 bool AudioProcessingImpl::was_stream_delay_set() const {
   // Used as callback from submodules, hence locking is not allowed.
   return capture_.was_stream_delay_set;
 }
 
 void AudioProcessingImpl::set_stream_key_pressed(bool key_pressed) {
   rtc::CritScope cs(&crit_capture_);
   capture_.key_pressed = key_pressed;
 }
 
+void AudioProcessingImpl::SetLevelControllerInitialLevel(float level) {
+  rtc::CritScope cs(&crit_capture_);
+  private_submodules_->level_controller->SetInitialLevel(level);
+}
+
 void AudioProcessingImpl::set_delay_offset_ms(int offset) {
   rtc::CritScope cs(&crit_capture_);
   capture_.delay_offset_ms = offset;
 }
 
 int AudioProcessingImpl::delay_offset_ms() const {
   rtc::CritScope cs(&crit_capture_);
   return capture_.delay_offset_ms;
 }
 
@@ skipping 487 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