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

webrtc/modules/audio_processing/audio_processing_impl.h

 /*
  *  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 64 matching lines @@
                     float* const* dest) override;
   int ProcessStream(const float* const* src,
                     const StreamConfig& input_config,
                     const StreamConfig& output_config,
                     float* const* dest) override;
   void set_output_will_be_muted(bool muted) override;
   int set_stream_delay_ms(int delay) override;
   void set_delay_offset_ms(int offset) override;
   int delay_offset_ms() const override;
   void set_stream_key_pressed(bool key_pressed) override;
+  void SetLevelControllerInitialLevel(float level) override;
 
   // Render-side exclusive methods possibly running APM in a
   // multi-threaded manner. Acquire the render lock.
   int ProcessReverseStream(AudioFrame* frame) override;
   int AnalyzeReverseStream(const float* const* data,
                            size_t samples_per_channel,
                            int sample_rate_hz,
                            ChannelLayout layout) override;
   int ProcessReverseStream(const float* const* src,
                            const StreamConfig& reverse_input_config,
@@ skipping 248 matching lines @@
 
   struct ApmRenderState {
     std::unique_ptr<AudioConverter> render_converter;
     std::unique_ptr<AudioBuffer> render_audio;
   } render_ GUARDED_BY(crit_render_);
 };
 
 }  // namespace webrtc
 
 #endif  // WEBRTC_MODULES_AUDIO_PROCESSING_AUDIO_PROCESSING_IMPL_H_