MediaStream audio object graph untangling and clean-ups.

content/renderer/media/webrtc_audio_capturer.h

Index: content/renderer/media/webrtc_audio_capturer.h
diff --git a/content/renderer/media/webrtc_audio_capturer.h b/content/renderer/media/webrtc_audio_capturer.h
index afa5f9e54b546f7a6b7b6211ccc2dd651525f189..c9d3bff55f455777c8b1f0c030212f41e113fdc5 100644
--- a/content/renderer/media/webrtc_audio_capturer.h
+++ b/content/renderer/media/webrtc_audio_capturer.h
@@ -12,10 +12,13 @@
 #include "base/files/file.h"
 #include "base/macros.h"
 #include "base/memory/ref_counted.h"
+#include "base/memory/scoped_ptr.h"
+#include "base/memory/weak_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"
@@ -41,8 +44,7 @@ class WebRtcLocalAudioTrack;
 // 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.
@@ -50,13 +52,15 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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
@@ -84,16 +88,9 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // 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;

[mcasas, 2016/02/26 01:28:19]

One thing that surprised me of audio processing
vs video is that the parameters of the audio stream
are passed partially in a preamble (as AudioParameters)
and partially with the AudioBus (inside AudioBus), and
there's still an overarching AudioBuffer that transports
all necessary data with each blob of audio, but a copy
is needed from AudioBus to AudioBuffer. Is AudioBuffer
the future? 

[miu, 2016/02/27 03:46:37]

I've never used AudioBuffer.  It's sole purpose seems to be to copy audio data
once, hold it for some time, and then copy it again.  That seems like an
anti-pattern to me, but perhaps there's a rationale I'm missing.

<rant>IMO, the purpose of copying data should be to share data, and only when it
would otherwise be impractical to guarantee the original data remains unchanged
while being shared.  Fast machines have allowed developers to become lazy w/ all
the copying and re-copying...</rant>

I'd rather see AudioBus become a little more like VideoFrame, where the audio
format is available as metadata alongside the audio samples.  Then, perhaps we
could all agree to use AudioBus everywhere and kill AudioBuffer and other
"solutions."

 
-  // 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.
@@ -110,10 +107,6 @@ class CONTENT_EXPORT WebRtcAudioCapturer
       const scoped_refptr<media::AudioCapturerSource>& source,
       media::AudioParameters params);
 
- protected:
-  friend class base::RefCountedThreadSafe<WebRtcAudioCapturer>;
-  ~WebRtcAudioCapturer() override;
-
  private:
   class TrackOwner;
   typedef TaggedList<TrackOwner> TrackList;
@@ -144,8 +137,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   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
@@ -176,7 +168,7 @@ class CONTENT_EXPORT WebRtcAudioCapturer
 
   // 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_;
 
@@ -205,6 +197,13 @@ class CONTENT_EXPORT WebRtcAudioCapturer
   // WebRtcAudioCapturer.
   MediaStreamAudioSource* const audio_source_;
 
+  // Used to calculate the signal level that shows in the UI.
+  MediaStreamAudioLevelCalculator level_calculator_;
+
+  // Provides weak pointers so that the stop callbacks given to
+  // WebRtcLocalAudioTracks can be canceled once they are no longer necessary.
+  base::WeakPtrFactory<WebRtcAudioCapturer> weak_factory_;
+
   DISALLOW_COPY_AND_ASSIGN(WebRtcAudioCapturer);
 };