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// 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_RENDERER_H_
#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
#include "base/memory/ref_counted.h"
#include "base/synchronization/lock.h"
#include "base/threading/thread_checker.h"
#include "content/renderer/media/webrtc_audio_device_impl.h"
#include "media/base/audio_decoder.h"
#include "media/base/audio_pull_fifo.h"
#include "media/base/audio_renderer_sink.h"
#include "webkit/media/media_stream_audio_renderer.h"
namespace media {
class AudioOutputDevice;
}
namespace content {
class WebRtcAudioRendererSource;
// This renderer handles calls from the pipeline and WebRtc ADM. It is used
// for connecting WebRtc MediaStream with the audio pipeline.
class CONTENT_EXPORT WebRtcAudioRenderer
: NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
NON_EXPORTED_BASE(public webkit_media::MediaStreamAudioRenderer) {
public:
explicit WebRtcAudioRenderer(int source_render_view_id);
// Initialize function called by clients like WebRtcAudioDeviceImpl.
// Stop() has to be called before |source| is deleted.
bool Initialize(WebRtcAudioRendererSource* source);
// Methods called by WebMediaPlayerMS and WebRtcAudioDeviceImpl.
// MediaStreamAudioRenderer implementation.
virtual void Start() OVERRIDE;
virtual void Play() OVERRIDE;
virtual void Pause() OVERRIDE;
virtual void Stop() OVERRIDE;
virtual void SetVolume(float volume) OVERRIDE;
virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
virtual bool IsLocalRenderer() const OVERRIDE;
protected:
virtual ~WebRtcAudioRenderer();
private:
enum State {
UNINITIALIZED,
PLAYING,
PAUSED,
};
// Used to DCHECK that we are called on the correct thread.
base::ThreadChecker thread_checker_;
// Flag to keep track the state of the renderer.
State state_;
// media::AudioRendererSink::RenderCallback implementation.
// These two methods are called on the AudioOutputDevice worker thread.
virtual int Render(media::AudioBus* audio_bus,
int audio_delay_milliseconds) OVERRIDE;
virtual void OnRenderError() OVERRIDE;
// Called by AudioPullFifo when more data is necessary.
// This method is called on the AudioOutputDevice worker thread.
void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus);
// The render view in which the audio is rendered into |sink_|.
const int source_render_view_id_;
// The sink (destination) for rendered audio.
scoped_refptr<media::AudioOutputDevice> sink_;
// Audio data source from the browser process.
WebRtcAudioRendererSource* source_;
// Buffers used for temporary storage during render callbacks.
// Allocated during initialization.
scoped_ptr<int16[]> buffer_;
// Protects access to |state_|, |source_| and |sink_|.
base::Lock lock_;
// Ref count for the MediaPlayers which are playing audio.
int play_ref_count_;
// Used to buffer data between the client and the output device in cases where
// the client buffer size is not the same as the output device buffer size.
scoped_ptr<media::AudioPullFifo> audio_fifo_;
// Contains the accumulated delay estimate which is provided to the WebRTC
// AEC.
int audio_delay_milliseconds_;
// Lengh of an audio frame in milliseconds.
double frame_duration_milliseconds_;
double fifo_io_ratio_;
DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
};
} // namespace content
#endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
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