Support multiple files for AEC dump.

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/files/file.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/aec_dump_message_filter.h"
 #include "content/renderer/media/webrtc_audio_capturer.h"
 #include "content/renderer/media/webrtc_audio_device_not_impl.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
@@ skipping 235 matching lines @@
   virtual ~WebRtcPlayoutDataSource() {}
 };
 
 // Note that this class inherits from webrtc::AudioDeviceModule but due to
 // the high number of non-implemented methods, we move the cruft over to the
 // WebRtcAudioDeviceNotImpl.
 class CONTENT_EXPORT WebRtcAudioDeviceImpl
     : NON_EXPORTED_BASE(public PeerConnectionAudioSink),
       NON_EXPORTED_BASE(public WebRtcAudioDeviceNotImpl),
       NON_EXPORTED_BASE(public WebRtcAudioRendererSource),
-      NON_EXPORTED_BASE(public WebRtcPlayoutDataSource) {
+      NON_EXPORTED_BASE(public WebRtcPlayoutDataSource),
+      NON_EXPORTED_BASE(public AecDumpMessageFilter::AecDumpObserver) {
  public:
   // The maximum volume value WebRtc uses.
   static const int kMaxVolumeLevel = 255;
 
   // Instances of this object are created on the main render thread.
   WebRtcAudioDeviceImpl();
 
   // webrtc::RefCountedModule implementation.
   // The creator must call AddRef() after construction and use Release()
   // to release the reference and delete this object.
@@ skipping 33 matching lines @@
 
   virtual int32_t MaxMicrophoneVolume(uint32_t* max_volume) const OVERRIDE;
   virtual int32_t MinMicrophoneVolume(uint32_t* min_volume) const OVERRIDE;
   virtual int32_t StereoPlayoutIsAvailable(bool* available) const OVERRIDE;
   virtual int32_t StereoRecordingIsAvailable(bool* available) const OVERRIDE;
   virtual int32_t PlayoutDelay(uint16_t* delay_ms) const OVERRIDE;
   virtual int32_t RecordingDelay(uint16_t* delay_ms) const OVERRIDE;
   virtual int32_t RecordingSampleRate(uint32_t* sample_rate) const OVERRIDE;
   virtual int32_t PlayoutSampleRate(uint32_t* sample_rate) const OVERRIDE;
 
+  // AecDumpMessageFilter::AecDumpObserver implementation.
+  virtual void OnAecDumpFile(
+      int id,
+      const IPC::PlatformFileForTransit& file_handle) OVERRIDE;
+  virtual void OnDisableAecDump() OVERRIDE;
+  virtual void OnIpcClosed() OVERRIDE;
+
   // Sets the |renderer_|, returns false if |renderer_| already exists.
   // Called on the main renderer thread.
   bool SetAudioRenderer(WebRtcAudioRenderer* renderer);
 
   // Adds/Removes the capturer to the ADM.
   // TODO(xians): Remove these two methods once the ADM does not need to pass
   // hardware information up to WebRtc.
   void AddAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
   void RemoveAudioCapturer(const scoped_refptr<WebRtcAudioCapturer>& capturer);
 
   // Gets the default capturer, which is the last capturer in |capturers_|.
   // The method can be called by both Libjingle thread and main render thread.
   scoped_refptr<WebRtcAudioCapturer> GetDefaultCapturer() const;
 
   // Gets paired device information of the capture device for the audio
   // renderer. This is used to pass on a session id, sample rate and buffer
   // size to a webrtc audio renderer (either local or remote), so that audio
   // will be rendered to a matching output device.
   // 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(base::File 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::map<int, scoped_refptr<WebRtcAudioCapturer> > CapturerMap;
   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.
 
   // Called on the AudioInputDevice worker thread.
   virtual int OnData(const int16* audio_data,
@@ skipping 17 matching lines @@
                           int audio_delay_milliseconds,
                           base::TimeDelta* current_time) 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();
+  // Registers capturer with ID |id| as an AEC dump consumer with the browser.
+  void RegisterAecDumpForId(int id);
+
+  // Disables the AEC dump on all capturers.
+  void DisableAecDump();
 
   // 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
+  // Map of captures which provides access to the native audio input layer
   // in the browser process.
-  CapturerList capturers_;
+  CapturerMap capturers_;
 
   // Provides access to the audio renderer in the browser process.
   scoped_refptr<WebRtcAudioRenderer> renderer_;
 
   // A list of raw pointer of WebRtcPlayoutDataSource::Sink objects which want
   // to get the playout data, the sink need to call RemovePlayoutSink()
   // before it goes away.
   PlayoutDataSinkList playout_sinks_;
 
   // Weak reference to the audio callback.
@@ skipping 20 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::File aec_dump_file_;
+  // Communication with browser for AEC dump.
+  // This is only used if audio track processing is enabled.
+  // TODO(xians): Remove the above comment line when option to disable audio
+  // track processing is removed.
+  scoped_refptr<AecDumpMessageFilter> aec_dump_message_filter_;
 
   // Flag to tell if audio processing is enabled in MediaStreamAudioProcessor.
   const bool is_audio_track_processing_enabled_;
 
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioDeviceImpl);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_DEVICE_IMPL_H_