Add chromium support for MediaStreamAudioDestinationNode

content/renderer/media/webaudio_capturer_source.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_WEBAUDIO_CAPTURER_SOURCE_H_
+#define CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_
+
+#include "base/memory/ref_counted.h"
+#include "base/synchronization/lock.h"
+#include "media/audio/audio_parameters.h"
+#include "media/base/audio_capturer_source.h"
+#include "media/base/audio_fifo.h"
+#include "third_party/WebKit/Source/Platform/chromium/public/WebAudioDestinationConsumer.h"
+#include "third_party/WebKit/Source/WebKit/chromium/public/platform/WebVector.h"
+
+namespace content {
+
+class WebRtcAudioCapturer;
+
+// WebAudioCapturerSource is the missing link between
+// WebAudio's MediaStreamAudioDestinationNode and WebRtcAudioCapturer.
+//
+// 1. WebKit calls the setFormat() method setting up the basic stream format
+//    (channels, and sample-rate).  At this time, it dispatches this information
+//    to the WebRtcAudioCapturer by calling its SetCapturerSource() method.
+// 2. Initialize() is called, where we should get back the same
+//    stream format information as (1).  We also get the CaptureCallback here.
+// 3. consumeAudio() is called periodically by WebKit which dispatches the
+//    audio stream to the CaptureCallback::Capture() method.
+class WebAudioCapturerSource
+    : public media::AudioCapturerSource,
+      public WebKit::WebAudioDestinationConsumer {
+ public:
+  explicit WebAudioCapturerSource(WebRtcAudioCapturer* capturer);
+
+  // WebAudioDestinationConsumer implementation.
+  // setFormat() is called early on, so that we can configure the capturer.
+  virtual void setFormat(size_t number_of_channels, float sample_rate) OVERRIDE;
+  // MediaStreamAudioDestinationNode periodically calls consumeAudio().
+  virtual void consumeAudio(const WebKit::WebVector<const float*>& audio_data,
+      size_t number_of_frames) OVERRIDE;
+
+  // AudioCapturerSource implementation.
+  virtual void Initialize(
+      const media::AudioParameters& params,
+      media::AudioCapturerSource::CaptureCallback* callback,
+      media::AudioCapturerSource::CaptureEventHandler* event_handler) OVERRIDE;
+
+  virtual void Start() OVERRIDE;
+  virtual void Stop() OVERRIDE;
+  virtual void SetVolume(double volume) OVERRIDE { }
+  virtual void SetDevice(int session_id) OVERRIDE { }
+  virtual void SetAutomaticGainControl(bool enable) OVERRIDE { }
+
+ private:
+  virtual ~WebAudioCapturerSource();
+
+  WebRtcAudioCapturer* capturer_;
+
+  int set_format_channels_;
+  media::AudioParameters params_;
+  media::AudioCapturerSource::CaptureCallback* callback_;
+
+  // Wraps data coming from HandleCapture().
+  scoped_ptr<media::AudioBus> wrapper_bus_;
+
+  // Bus for reading from FIFO and calling the CaptureCallback.
+  scoped_ptr<media::AudioBus> capture_bus_;
+
+  // Handles mismatch between WebAudio buffer size and WebRTC.
+  scoped_ptr<media::AudioFifo> fifo_;
+
+  // Synchronizes HandleCapture() with AudioCapturerSource calls.
+  base::Lock lock_;
+  bool started_;
+
+  DISALLOW_COPY_AND_ASSIGN(WebAudioCapturerSource);
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBAUDIO_CAPTURER_SOURCE_H_