Support the Aec dump for the APM in chrome

content/renderer/media/webrtc_audio_capturer.h

 // Copyright (c) 2012 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_CAPTURER_H_
 #define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_
 
 #include <list>
 #include <string>
 
 #include "base/callback.h"
 #include "base/memory/ref_counted.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/media/media_stream_options.h"
 #include "content/renderer/media/tagged_list.h"
 #include "media/audio/audio_input_device.h"
 #include "media/base/audio_capturer_source.h"
 #include "third_party/WebKit/public/platform/WebMediaConstraints.h"
 
 namespace media {
@@ skipping 78 matching lines @@
   // This method is exposed to the public because the media stream track can
   // call Stop() on its source.
   void Stop();
 
   // Called by the WebAudioCapturerSource to get the audio processing params.
   // This function is triggered by provideInput() on the WebAudio audio thread,
   // TODO(xians): Remove after moving APM from WebRtc to Chrome.
   void GetAudioProcessingParams(base::TimeDelta* delay, int* volume,
                                 bool* key_pressed);
 
-  // Use by the unittests to inject their own source to the capturer.
+  // Used by the unittests to inject their own source to the capturer.
   void SetCapturerSourceForTesting(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
+  void StartAecDump(const base::PlatformFile& aec_dump_file);
+  void StopAecDump();
+
  protected:
   friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
   virtual ~WebRtcAudioCapturer();
 
  private:
   class TrackOwner;
   typedef TaggedList<TrackOwner> TrackList;
 
   WebRtcAudioCapturer(int render_view_id,
                       const StreamDeviceInfo& device_info,
@@ skipping 12 matching lines @@
   // view id and device information. Return true if success, otherwise false.
   bool Initialize();
 
   // SetCapturerSource() is called if the client on the source side desires to
   // provide their own captured audio data. Client is responsible for calling
   // Start() on its own source to have the ball rolling.
   // Called on the main render thread.
   void SetCapturerSource(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::ChannelLayout channel_layout,
-      float sample_rate,
-      int effects,
-      const blink::WebMediaConstraints& constraints);
+      float sample_rate);

[Henrik Grunell, 2014/03/06 19:55:12]

What has removing those parameters to do with the AEC dump?

[no longer working on chromium, 2014/03/06 20:00:36]

They are used to create the MediaStreamAudioProcessor.
they are not needed any more since MediaStreamAudioProcessor will be created on
the constructor.

 
   // Starts recording audio.
   // Triggered by AddSink() on the main render thread or a Libjingle working
   // thread. It should NOT be called under |lock_|.
   void Start();
 
   // Helper function to get the buffer size based on |peer_connection_mode_|
   // and sample rate;
   int GetBufferSize(int sample_rate) const;
 
@@ skipping 43 matching lines @@
   // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
   // of RenderThread.
   WebRtcAudioDeviceImpl* audio_device_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_