Support the Aec dump for the APM in chrome

content/renderer/media/webrtc_audio_device_impl.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_WEBRTC_AUDIO_DEVICE_IMPL_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_
 
 #include <string>
 #include <vector>
 
 #include "base/basictypes.h"
 #include "base/compiler_specific.h"
 #include "base/logging.h"
 #include "base/memory/ref_counted.h"
 #include "base/memory/scoped_ptr.h"
 #include "base/threading/thread_checker.h"
 #include "content/common/content_export.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_not_impl.h"
 #include "content/renderer/media/webrtc_audio_renderer.h"
+#include "ipc/ipc_platform_file.h"
 #include "media/base/audio_capturer_source.h"
 #include "media/base/audio_renderer_sink.h"
 
 // A WebRtcAudioDeviceImpl instance implements the abstract interface
 // webrtc::AudioDeviceModule which makes it possible for a user (e.g. webrtc::
 // VoiceEngine) to register this class as an external AudioDeviceModule (ADM).
 // Then WebRtcAudioDeviceImpl::SetSessionId() needs to be called to set the
 // session id that tells which device to use. The user can then call
 // WebRtcAudioDeviceImpl::StartPlayout() and
 // WebRtcAudioDeviceImpl::StartRecording() from the render process to initiate
@@ skipping 313 matching lines @@
   // Returns true if the capture device has a paired output device, otherwise
   // false. Note that if there are more than one open capture device the
   // function will not be able to pick an appropriate device and return false.
   bool GetAuthorizedDeviceInfoForAudioRenderer(
       int* session_id, int* output_sample_rate, int* output_buffer_size);
 
   const scoped_refptr<WebRtcAudioRenderer>& renderer() const {
     return renderer_;
   }
 
+  // Enables the Aec dump.  If the default capturer exists, it will call
+  // StartAecDump() on the capturer and pass the ownership of the file to
+  // WebRtc. Otherwise it will hold the file until a capturer is added.
+  void EnableAecDump(const base::PlatformFile& aec_dump_file);
+
+  // Disables the Aec dump.  When this method is called, the ongoing Aec dump
+  // on WebRtc will be stopped.
+  void DisableAecDump();
+
  private:
   typedef std::list<scoped_refptr<WebRtcAudioCapturer> > CapturerList;
   typedef std::list<WebRtcPlayoutDataSource::Sink*> PlayoutDataSinkList;
   class RenderBuffer;
 
   // Make destructor private to ensure that we can only be deleted by Release().
   virtual ~WebRtcAudioDeviceImpl();
 
   // PeerConnectionAudioSink implementation.
 
@@ skipping 18 matching lines @@
                           int sample_rate,
                           int audio_delay_milliseconds) OVERRIDE;
 
   // Called on the main render thread.
   virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;
 
   // WebRtcPlayoutDataSource implementation.
   virtual void AddPlayoutSink(WebRtcPlayoutDataSource::Sink* sink) OVERRIDE;
   virtual void RemovePlayoutSink(WebRtcPlayoutDataSource::Sink* sink) OVERRIDE;
 
+  // Helper to start the Aec dump if the default capturer exists.
+  void MaybeStartAecDump();
+
   // Used to DCHECK that we are called on the correct thread.
   base::ThreadChecker thread_checker_;
 
   int ref_count_;
 
   // List of captures which provides access to the native audio input layer
   // in the browser process.
   CapturerList capturers_;
 
   // Provides access to the audio renderer in the browser process.
@@ skipping 28 matching lines @@
   bool recording_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   uint32_t microphone_volume_;
 
   // Buffer used for temporary storage during render callback.
   // It is only accessed by the audio render thread.
   std::vector<int16> render_buffer_;
 
+  // Used for start the Aec dump on the default capturer.
+  base::PlatformFile aec_dump_file_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_