Support the Aec dump for the APM in chrome

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/platform_file.h"
 #include "base/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
 #include "third_party/libjingle/source/talk/app/webrtc/mediastreaminterface.h"
 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 #include "third_party/webrtc/modules/interface/module_common_types.h"
 
@@ skipping 22 matching lines @@
 // processing components like AGC, AEC and NS. It enables the components based
 // on the getUserMedia constraints, processes the data and outputs it in a unit
 // of 10 ms data chunk.
 class CONTENT_EXPORT MediaStreamAudioProcessor :
     NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
     NON_EXPORTED_BASE(public AudioProcessorInterface) {
  public:
   // |playout_data_source| is used to register this class as a sink to the
   // WebRtc playout data for processing AEC. If clients do not enable AEC,
   // |playout_data_source| won't be used.
-  MediaStreamAudioProcessor(const media::AudioParameters& source_params,
-                            const blink::WebMediaConstraints& constraints,
+  MediaStreamAudioProcessor(const blink::WebMediaConstraints& constraints,
                             int effects,
                             WebRtcPlayoutDataSource* playout_data_source);
 
+  // Called when format of the capture data has changed.
+  // Called on the main render thread.  The caller is responsible for stopping
+  // the capture thread before calling this method.
+  // After this method, the capture thread will be changed to a new capture
+  // thread.
+  void OnCaptureFormatChanged(const media::AudioParameters& source_params);
+
   // Pushes capture data in |audio_source| to the internal FIFO.
   // Called on the capture audio thread.
   void PushCaptureData(media::AudioBus* audio_source);
 
   // 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| receives the new microphone volume from the AGC.
   // The new microphoen volume range is [0, 255], and the value will be 0 if
   // the microphone volume should not be adjusted.
   // 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; }
 
+  // Starts/Stops the Aec dump on the |audio_processing_|.
+  // Called on the main render thread.
+  // This method takes the ownership of |aec_dump_file|.
+  void StartAecDump(const base::PlatformFile& aec_dump_file);
+  void StopAecDump();
+
  protected:
   friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>;
   virtual ~MediaStreamAudioProcessor();
 
  private:
   friend class MediaStreamAudioProcessorTest;
 
   class MediaStreamAudioConverter;
 
   // WebRtcPlayoutDataSource::Sink implementation.
@@ skipping 74 matching lines @@
 
   // This flag is used to show the result of typing detection.
   // It can be accessed by the capture audio thread and by the libjingle thread
   // which calls GetStats().
   base::subtle::Atomic32 typing_detected_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_