Feed the render data to MediaStreamAudioProcessor and used AudioBus in render callback

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/synchronization/lock.h"
 #include "base/threading/thread_checker.h"
 #include "base/time/time.h"
 #include "content/common/content_export.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
 #include "media/base/audio_converter.h"
 #include "third_party/webrtc/modules/audio_processing/include/audio_processing.h"
 #include "third_party/webrtc/modules/interface/module_common_types.h"
 
 namespace blink {
 class WebMediaConstraints;
 }
 
 namespace media {
 class AudioBus;
 class AudioFifo;
 class AudioParameters;
 }  // namespace media
 
 namespace webrtc {
 class AudioFrame;
 }
 
 namespace content {
 
 class RTCMediaConstraints;
 
 // 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 :
-    public base::RefCountedThreadSafe<MediaStreamAudioProcessor> {
+    public base::RefCountedThreadSafe<MediaStreamAudioProcessor>,
+    public WebRtcAudioRendererSource {
  public:
   MediaStreamAudioProcessor(const media::AudioParameters& source_params,
                             const blink::WebMediaConstraints& constraints,
-                            int effects);
+                            int effects,
+                            WebRtcAudioDeviceImpl* audio_device);

[miu, 2014/01/24 22:27:19]

Please comment this ctor to note that |audio_device| is optional (and what
behavior does that change?).

[miu, 2014/01/24 22:27:19]

Also, should you be using WebRtcAudioRendererSource* or WebRtcAudioDeviceImpl*
as the pointer type here?  It looks like the interface type would be sufficient.

[no longer working on chromium, 2014/01/27 17:09:33]

Done with adding a comment to explain what |audio_device| is used for.

|audio_device| is used to register getting the webrtc render data for AEC. So it
won't NULL for production code, but for some unittests, it can be NULL.

[no longer working on chromium, 2014/01/27 17:09:33]

audio_device_->AddRenderDataObserver() and
audio_device_->RemoveRenderDataObserver()
are not interface to WebRtcAudioRendererSource, but methods to
WebRtcAudioDeviceImpl. Do you want to me change them into interfaces?

 
   // Pushes capture data in |audio_source| to the internal FIFO.
   // Called on the capture audio thread.
   void PushCaptureData(media::AudioBus* audio_source);
 
-  // Push the render audio to webrtc::AudioProcessing for analysis. This is
-  // needed iff echo processing is enabled.
-  // |render_audio| is the pointer to the render audio data, its format
-  // is specified by |sample_rate|, |number_of_channels| and |number_of_frames|.
-  // Called on the render audio thread.
-  void PushRenderData(const int16* render_audio,
-                      int sample_rate,
-                      int number_of_channels,
-                      int number_of_frames,
-                      base::TimeDelta render_delay);
-
   // Processes a block of 10 ms data from the internal FIFO and outputs it via
   // |out|. |out| is the address of the pointer that will be pointed to
   // the post-processed data if the method is returning a true. The lifetime
   // of the data represeted by |out| is guaranteed to outlive the method call.
   // That also says *|out| won't change until this method is called again.
   // Returns true if the internal FIFO has at least 10 ms data for processing,
   // otherwise false.
   // |capture_delay|, |volume| and |key_pressed| will be passed to
   // webrtc::AudioProcessing to help processing the data.
   // Called on the capture audio thread.
@@ skipping 12 matching lines @@
   // Accessor to check if the audio processing is enabled or not.
   bool has_audio_processing() const { return audio_processing_ != NULL; }
 
  protected:
   friend class base::RefCountedThreadSafe<MediaStreamAudioProcessor>;
   virtual ~MediaStreamAudioProcessor();
 
  private:
   class MediaStreamAudioConverter;
 
+  // WebRtcAudioRendererSource implementation.
+  virtual void RenderData(media::AudioBus* audio_bus,
+                          int sample_rate,
+                          int audio_delay_milliseconds) OVERRIDE;
+  virtual void RemoveAudioRenderer(WebRtcAudioRenderer* renderer) OVERRIDE;
+
   // Helper to initialize the WebRtc AudioProcessing.
   void InitializeAudioProcessingModule(
       const blink::WebMediaConstraints& constraints, int effects);
 
   // Helper to initialize the capture converter.
   void InitializeCaptureConverter(const media::AudioParameters& source_params);
 
   // Helper to initialize the render converter.
   void InitializeRenderConverterIfNeeded(int sample_rate,
                                          int number_of_channels,
@@ skipping 25 matching lines @@
   // Converter used for the down-mixing and resampling of the render data when
   // the AEC is enabled.
   scoped_ptr<MediaStreamAudioConverter> render_converter_;
 
   // AudioFrame used to hold the output of |render_converter_|.
   webrtc::AudioFrame render_frame_;
 
   // Data bus to help converting interleaved data to an AudioBus.
   scoped_ptr<media::AudioBus> render_data_bus_;
 
-  // Used to DCHECK that some methods are called on the main render thread.
-  base::ThreadChecker main_thread_checker_;
+  // Raw pointer to the WebRtcAudioDeviceImpl, which is valid for the lifetime
+  // of RenderThread.
+  WebRtcAudioDeviceImpl* audio_device_;

[miu, 2014/01/24 22:27:19]

nit: Should be:

  WebRtcAudioDeviceImpl* const audio_device_;

Since the member is never changed after construction.

[no longer working on chromium, 2014/01/27 17:09:33]

Done.

 
   // Used to DCHECK that some methods are called on the capture audio thread.
   base::ThreadChecker capture_thread_checker_;
 
   // Used to DCHECK that PushRenderData() is called on the render audio thread.
   base::ThreadChecker render_thread_checker_;
 };
 
 }  // namespace content
 
 #endif  // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_PROCESSOR_H_