Wire up AGC to the MediaStreamAudioProcessor

content/renderer/media/media_stream_audio_processor.h

 // Copyright 2013 The Chromium Authors. All rights reserved.
 // Use of this source code is governed by a BSD-style license that can be
 // found in the LICENSE file.
 
 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_
 
 #include "base/atomicops.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
@@ skipping 45 matching lines @@
                       int sample_rate,
                       int number_of_channels,
                       int number_of_frames,
                       base::TimeDelta render_delay);
 
   // Processes a block of 10 ms data from the internal FIFO and outputs it via
   // |out|. |out| is the address of the pointer that will be pointed to
   // the post-processed data if the method is returning a true. The lifetime
   // of the data represeted by |out| is guaranteed to outlive the method call.
   // That also says *|out| won't change until this method is called again.
+  // |new_volume| contains the new microphone volume from the AGC, the value
+  // will be 0 if the microphone volume is not adjusted.

[tommi (sloooow) - chröme, 2014/01/21 15:06:20]

nit:
// |new_volume| receives the new microphone volume from the AGC.
// The value will be 0 if the microphone volume should not be
// adjusted.

do you want to add to that a note about the range?

[no longer working on chromium, 2014/01/23 12:46:08]

Done.

   // Returns true if the internal FIFO has at least 10 ms data for processing,
   // otherwise false.
   // |capture_delay|, |volume| and |key_pressed| will be passed to
   // webrtc::AudioProcessing to help processing the data.
   // Called on the capture audio thread.
   bool ProcessAndConsumeData(base::TimeDelta capture_delay,
                              int volume,
                              bool key_pressed,
+                             int* new_volume,
                              int16** out);
 
 
   // The audio format of the input to the processor.
   const media::AudioParameters& InputFormat() const;
 
   // The audio format of the output from the processor.
   const media::AudioParameters& OutputFormat() const;
 
   // Accessor to check if the audio processing is enabled or not.
   bool has_audio_processing() const { return audio_processing_ != NULL; }
 
  protected:
   friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>;
   virtual ~MediaStreamAudioProcessor();
 
  private:
+  friend class MediaStreamAudioProcessorTest;
+
   class MediaStreamAudioConverter;
 
   // Helper to initialize the WebRtc AudioProcessing.
   void InitializeAudioProcessingModule(
       const blink::WebMediaConstraints& constraints, int effects);
 
   // Helper to initialize the capture converter.
   void InitializeCaptureConverter(const media::AudioParameters& source_params);
 
   // Helper to initialize the render converter.
   void InitializeRenderConverterIfNeeded(int sample_rate,
                                          int number_of_channels,
                                          int frames_per_buffer);
 
   // Called by ProcessAndConsumeData().
-  void ProcessData(webrtc::AudioFrame* audio_frame,
-                   base::TimeDelta capture_delay,
-                   int volume,
-                   bool key_pressed);
+  // Returns the new microphone volume in the range of |0, 255].
+  // When the volume does not need to be updated, it returns 0.
+  int ProcessData(webrtc::AudioFrame* audio_frame,
+                  base::TimeDelta capture_delay,
+                  int volume,
+                  bool key_pressed);
 
   // Called when the processor is going away.
   void StopAudioProcessing();
 
   // Cached value for the render delay latency. This member is accessed by
   // both the capture audio thread and the render audio thread.
   base::subtle::Atomic32 render_delay_ms_;
 
   // webrtc::AudioProcessing module which does AEC, AGC, NS, HighPass filter,
   // ..etc.
@@ skipping 16 matching lines @@
   scoped_ptr<media::AudioBus> render_data_bus_;
 
   // Used to DCHECK that some methods are called on the main render thread.
   base::ThreadChecker main_thread_checker_;
 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   // Used to DCHECK that PushRenderData() is called on the render audio thread.
   base::ThreadChecker render_thread_checker_;
+
+  // Flag to enable the stereo channels swapping.
+  bool stereo_channels_swapping_;

[tommi (sloooow) - chröme, 2014/01/21 15:06:20]

nit: audio_mirroring_

[no longer working on chromium, 2014/01/23 12:46:08]

Done.

 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_