Switch LiveAudio to source provider solution.

content/renderer/media/webrtc_local_audio_source_provider.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_LOCAL_AUDIO_SOURCE_PROVIDER_H_
+#define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_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/content_export.h"
+#include "media/base/audio_converter.h"
+#include "third_party/WebKit/public/platform/WebVector.h"
+#include "third_party/WebKit/public/web/WebAudioSourceProvider.h"
+
+namespace media {
+class AudioBus;
+class AudioConverter;
+class AudioFifo;
+class AudioParameters;
+}
+
+namespace WebKit {
+class WebAudioSourceProviderClient;
+}
+
+namespace content {
+
+// WebRtcLocalAudioSourceProvider provides a bridge between classes:
+//     WebRtcAudioCapturer ---> WebKit::WebAudioSourceProvider
+//
+// WebRtcLocalAudioSourceProvider works as a sink to the WebRtcAudiocapturer
+// and store the capture data to a FIFO. When the media stream is connected to
+// WebAudio as a source provider, WebAudio will periodically call
+// provideInput() to get the data from the FIFO.
+//
+// All calls are protected by a lock.
+class CONTENT_EXPORT WebRtcLocalAudioSourceProvider
+    : NON_EXPORTED_BASE(public media::AudioConverter::InputCallback),
+      NON_EXPORTED_BASE(public WebKit::WebAudioSourceProvider) {
+ public:
+  static const size_t kWebAudioRenderBufferSize;
+
+  WebRtcLocalAudioSourceProvider();
+  virtual ~WebRtcLocalAudioSourceProvider();
+
+  // Initialize function for the souce provider. This can be called multiple
+  // times if the source format has changed.
+  void Initialize(const media::AudioParameters& source_params);
+
+  // Called by the WebRtcAudioCapturer to deliever captured data into fifo on
+  // the capture audio thread.
+  void DeliverData(media::AudioBus* audio_source,
+                   int audio_delay_milliseconds,
+                   int volume,
+                   bool key_pressed);
+
+  // Called by the WebAudioCapturerSource to get the audio processing params.
+  // This function is triggered by provideInput() on the WebAudio audio thread,
+  // so it has been under the protection of |lock_|.
+  void GetAudioProcessingParams(int* delay_ms, int* volume, bool* key_pressed);
+
+  // WebKit::WebAudioSourceProvider implementation.
+  virtual void setClient(WebKit::WebAudioSourceProviderClient* client) OVERRIDE;
+  virtual void provideInput(const WebKit::WebVector<float*>& audio_data,
+                            size_t number_of_frames) OVERRIDE;
+
+  // media::AudioConverter::Inputcallback implementation.
+  // This function is triggered by provideInput()on the WebAudio audio thread,
+  // so it has been under the protection of |lock_|.
+  virtual double ProvideInput(media::AudioBus* audio_bus,
+                              base::TimeDelta buffer_delay) OVERRIDE;
+
+  // Method to allow the unittests to inject its own sink parameters to avoid
+  // query the hardware.
+  // TODO(xians,tommi): Remove and instead offer a way to inject the sink
+  // parameters so that the implementation doesn't rely on the global default
+  // hardware config but instead gets the parameters directly from the sink
+  // (WebAudio in this case). Ideally the unit test should be able to use that
+  // same mechanism to inject the sink parameters for testing.
+  void SetSinkParamsForTesting(const media::AudioParameters& sink_params);
+
+ private:
+  // Used to DCHECK that we are called on the correct thread.
+  base::ThreadChecker thread_checker_;
+
+  scoped_ptr<media::AudioConverter> audio_converter_;
+  scoped_ptr<media::AudioFifo> fifo_;
+  scoped_ptr<media::AudioBus> bus_wrapper_;
+  int audio_delay_ms_;
+  int volume_;
+  bool key_pressed_;
+  bool is_enabled_;
+  media::AudioParameters source_params_;
+  media::AudioParameters sink_params_;
+
+  // Protects all the member variables above.
+  base::Lock lock_;
+
+  // Used to report the correct delay to |webaudio_source_|.
+  base::TimeTicks last_fill_;
+
+  DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioSourceProvider);
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_SOURCE_PROVIDER_H_