Split out audio debug recording from RenderProcessHostImpl.

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/files/file.h"
 #include "base/gtest_prod_util.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/public/common/media_stream_request.h"
-#include "content/renderer/media/aec_dump_message_filter.h"
+#include "content/renderer/media/audio_debug_recorder.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"
 
 namespace blink {
 class WebMediaConstraints;
 }
 
 namespace media {
@@ skipping 12 matching lines @@
 class MediaStreamAudioBus;
 class MediaStreamAudioFifo;
 class RTCMediaConstraints;
 
 using webrtc::AudioProcessorInterface;
 
 // This class owns an object of webrtc::AudioProcessing which contains signal
 // 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),
-    NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpDelegate) {
+class CONTENT_EXPORT MediaStreamAudioProcessor
+    : NON_EXPORTED_BASE(public WebRtcPlayoutDataSource::Sink),
+      NON_EXPORTED_BASE(public AudioProcessorInterface),
+      NON_EXPORTED_BASE(public AudioDebugRecorder::AecDumpDelegate) {
  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 blink::WebMediaConstraints& constraints,
       const MediaStreamDevice::AudioDeviceParameters& input_params,
       WebRtcPlayoutDataSource* playout_data_source);
 
   // Called when the format of the capture data has changed.
@@ skipping 34 matching lines @@
   // The audio formats of the capture input to and output from the processor.
   // Must only be called on the main render or audio capture threads.
   const media::AudioParameters& InputFormat() const;
   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; }
 
   // AecDumpMessageFilter::AecDumpDelegate implementation.
   // Called on the main render thread.
-  void OnAecDumpFile(const IPC::PlatformFileForTransit& file_handle) override;
+  void OnAecDumpFile(base::PlatformFile file_handle) override;
   void OnDisableAecDump() override;
   void OnIpcClosing() override;
 
  protected:
   ~MediaStreamAudioProcessor() override;
 
  private:
   friend class MediaStreamAudioProcessorTest;
   FRIEND_TEST_ALL_PREFIXES(MediaStreamAudioProcessorTest,
-                           GetAecDumpMessageFilter);
+                           GetAudioDebugRecorder);
 
   // WebRtcPlayoutDataSource::Sink implementation.
   void OnPlayoutData(media::AudioBus* audio_bus,
                      int sample_rate,
                      int audio_delay_milliseconds) override;
   void OnPlayoutDataSourceChanged() override;
 
   // webrtc::AudioProcessorInterface implementation.
   // This method is called on the libjingle thread.
   void GetStats(AudioProcessorStats* stats) override;
@@ skipping 58 matching lines @@
   // Flag to enable stereo channel mirroring.
   bool audio_mirroring_;
 
   scoped_ptr<webrtc::TypingDetection> typing_detector_;
   // 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_;
 
   // Communication with browser for AEC dump.
-  scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
+  scoped_refptr<AudioDebugRecorder> audio_debug_recorder_;
 
   // Flag to avoid executing Stop() more than once.
   bool stopped_;
 
   // Object for logging echo information when the AEC is enabled. Accessible by
   // the libjingle thread through GetStats().
   scoped_ptr<EchoInformation> echo_information_;
 
   DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioProcessor);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_