MediaStream audio object graph untangling and clean-ups.

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/files/file.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
+#include "base/memory/scoped_ptr.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/media_stream_audio_level_calculator.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 {
 class AudioBus;
 }
 
 namespace content {
 
 class MediaStreamAudioProcessor;
 class MediaStreamAudioSource;
 class WebRtcAudioDeviceImpl;
 class WebRtcLocalAudioRenderer;
 class WebRtcLocalAudioTrack;
 
 // This class manages the capture data flow by getting data from its
 // |source_|, and passing it to its |tracks_|.
 // The threading model for this class is rather complex since it will be
 // created on the main render thread, captured data is provided on a dedicated
 // AudioInputDevice thread, and methods can be called either on the Libjingle
 // thread or on the main render thread but also other client threads
 // if an alternative AudioCapturerSource has been set.
 class CONTENT_EXPORT WebRtcAudioCapturer
-    : public base::RefCountedThreadSafe<WebRtcAudioCapturer>,
-      NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
+    : NON_EXPORTED_BASE(public media::AudioCapturerSource::CaptureCallback) {
  public:
   // Used to construct the audio capturer. |render_frame_id| specifies the
   // RenderFrame consuming audio for capture; -1 is used for tests.
   // |device_info| contains all the device information that the capturer is
   // created for. |constraints| contains the settings for audio processing.
   // TODO(xians): Implement the interface for the audio source and move the
   // |constraints| to ApplyConstraints(). Called on the main render thread.
-  static scoped_refptr<WebRtcAudioCapturer> CreateCapturer(
+  static scoped_ptr<WebRtcAudioCapturer> CreateCapturer(
       int render_frame_id,
       const StreamDeviceInfo& device_info,
       const blink::WebMediaConstraints& constraints,
       WebRtcAudioDeviceImpl* audio_device,
       MediaStreamAudioSource* audio_source);
 
+  ~WebRtcAudioCapturer() override;
+
   // Add a audio track to the sinks of the capturer.
   // WebRtcAudioDeviceImpl calls this method on the main render thread but
   // other clients may call it from other threads. The current implementation
   // does not support multi-thread calling.
   // The first AddTrack will implicitly trigger the Start() of this object.
   void AddTrack(WebRtcLocalAudioTrack* track);
 
   // Remove a audio track from the sinks of the capturer.
   // If the track has been added to the capturer, it  must call RemoveTrack()
   // before it goes away.
   // Called on the main render thread or libjingle working thread.
   void RemoveTrack(WebRtcLocalAudioTrack* track);
 
   // Called when a stream is connecting to a peer connection. This will set
   // up the native buffer size for the stream in order to optimize the
   // performance for peer connection.
   void EnablePeerConnectionMode();
 
   // Volume APIs used by WebRtcAudioDeviceImpl.
   // Called on the AudioInputDevice audio thread.
   void SetVolume(int volume);
   int Volume() const;
   int MaxVolume() const;
 
   // Audio parameters utilized by the source of the audio capturer.
   // TODO(phoglund): Think over the implications of this accessor and if we can
   // remove it.
-  media::AudioParameters source_audio_parameters() const;
+  media::AudioParameters GetInputFormat() const;
 
-  // Gets information about the paired output device. Returns true if such a
-  // device exists.
-  bool GetPairedOutputParameters(int* session_id,
-                                 int* output_sample_rate,
-                                 int* output_frames_per_buffer) const;
-
-  const std::string& device_id() const { return device_info_.device.id; }
-  int session_id() const { return device_info_.session_id; }
+  const StreamDeviceInfo& device_info() const { return device_info_; }
 
   // Stops recording audio. This method will empty its track lists since
   // stopping the capturer will implicitly invalidate all its tracks.
   // This method is exposed to the public because the MediaStreamAudioSource can
   // call Stop()
   void Stop();
 
   // Returns the output format.
   // Called on the main render thread.
   media::AudioParameters GetOutputFormat() const;
 
   // Used by clients to inject their own source to the capturer.
   void SetCapturerSource(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
- protected:
-  friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
-  ~WebRtcAudioCapturer() override;
-
  private:
   class TrackOwner;
   typedef TaggedList<TrackOwner> TrackList;
 
   WebRtcAudioCapturer(int render_frame_id,
                       const StreamDeviceInfo& device_info,
                       const blink::WebMediaConstraints& constraints,
                       WebRtcAudioDeviceImpl* audio_device,
                       MediaStreamAudioSource* audio_source);
 
@@ skipping 10 matching lines @@
   bool Initialize();
 
   // SetCapturerSourceInternal() 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 get the ball rolling.
   // Called on the main render thread.
   // buffer_size is optional. Set to 0 to let it be chosen automatically.
   void SetCapturerSourceInternal(
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::ChannelLayout channel_layout,
-      int sample_rate,
-      int buffer_size);
+      int sample_rate);
 
   // 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 10 matching lines @@
   TrackList tracks_;
 
   // The audio data source from the browser process.
   scoped_refptr<media::AudioCapturerSource> source_;
 
   // Cached audio constraints for the capturer.
   blink::WebMediaConstraints constraints_;
 
   // Audio processor doing processing like FIFO, AGC, AEC and NS. Its output
   // data is in a unit of 10 ms data chunk.
-  scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
+  const scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
 
   bool running_;
 
   int render_frame_id_;
 
   // Cached information of the device used by the capturer.
   const StreamDeviceInfo device_info_;
 
   // Stores latest microphone volume received in a CaptureData() callback.
   // Range is [0, 255].
   int volume_;
 
   // Flag which affects the buffer size used by the capturer.
   bool peer_connection_mode_;
 
   // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
   // of RenderThread.
   WebRtcAudioDeviceImpl* audio_device_;
 
   // Raw pointer to the MediaStreamAudioSource object that holds a reference
   // to this WebRtcAudioCapturer.
   // Since |audio_source_| is owned by a blink::WebMediaStreamSource object and
   // blink guarantees that the blink::WebMediaStreamSource outlives any
   // blink::WebMediaStreamTrack connected to the source, |audio_source_| is
   // guaranteed to exist as long as a WebRtcLocalAudioTrack is connected to this
   // WebRtcAudioCapturer.
   MediaStreamAudioSource* const audio_source_;
 
+  // Used to calculate the signal level that shows in the UI.
+  MediaStreamAudioLevelCalculator level_calculator_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_