Adding audio support to the new webmediaplyer

content/renderer/media/webrtc_audio_renderer.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 "base/memory/ref_counted.h"
+#include "base/synchronization/lock.h"
+#include "content/renderer/media/webrtc_audio_device_impl.h"
+#include "media/base/audio_decoder.h"
+#include "media/base/audio_renderer_sink.h"
+#include "media/base/rtc_audio_renderer.h"
+
+namespace content {
+
+class WebRtcAudioRendererSource;
+
+// This renderer handles calls from the pipeline and WebRtc ADM. It is used
+// for connecting WebRtc MediaStream with pipeline.
+class WebRtcAudioRenderer
+    : public media::RtcAudioRenderer,
+      NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback) {
+ public:
+  WebRtcAudioRenderer();
+
+  // Initialize function called by clients like WebRtcAudioDeviceImpl. Note,
+  // Stop() has to be called before |source| is deleted.
+  void Initialize(WebRtcAudioRendererSource* source);
+
+  // Methods called by WebMediaPlayerMS and WebRtcAudioDeviceImpl.
+  // RtcAudioRenderer implementation.
+  virtual void Play() OVERRIDE;
+  virtual void Pause() OVERRIDE;
+  virtual void Stop() OVERRIDE;
+  virtual void SetVolume(float volume) OVERRIDE;
+
+ protected:
+  virtual ~WebRtcAudioRenderer();
+
+ private:
+  enum State {
+    UNINITIALIZED,
+    PLAYING,
+    PAUSED,
+  };
+  // Flag to keep track the state of the renderer.
+  State state_;
+
+  // media::AudioRendererSink::RenderCallback implementation.
+  virtual int Render(media::AudioBus* audio_bus,
+                     int audio_delay_milliseconds) OVERRIDE;
+  virtual void OnRenderError() OVERRIDE;
+
+  // The sink (destination) for rendered audio.
+  scoped_refptr<media::AudioRendererSink> sink_;
+
+  // Audio data source from the browser process.
+  WebRtcAudioRendererSource* source_;
+
+  // Cached values of utilized audio parameters. Platform dependent.
+  media::AudioParameters params_;
+
+  // Buffers used for temporary storage during render callbacks.
+  // Allocated during initialization.
+  scoped_array<int16> buffer_;
+
+  // Protect access to |state_|.
+  base::Lock lock_;
+
+  DISALLOW_COPY_AND_ASSIGN(WebRtcAudioRenderer);
+};
+
+}  // namespace content
+
+#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_CAPTURER_H_