// 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/non_thread_safe.h"
#include "base/threading/thread_checker.h"
#include "content/renderer/media/media_stream_audio_renderer.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 "media/base/channel_layout.h"
namespace media {
class AudioOutputDevice;
} // namespace media
namespace webrtc {
class AudioSourceInterface;
class MediaStreamInterface;
} // namespace webrtc
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 MediaStreamAudioRenderer) {
public:
// This is a little utility class that holds the configured state of an audio
// stream.
// It is used by both WebRtcAudioRenderer and SharedAudioRenderer (see cc
// file) so a part of why it exists is to avoid code duplication and track
// the state in the same way in WebRtcAudioRenderer and SharedAudioRenderer.
class PlayingState : public base::NonThreadSafe {
public:
PlayingState() : playing_(false), volume_(1.0f) {}
bool playing() const {
DCHECK(CalledOnValidThread());
return playing_;
}
void set_playing(bool playing) {
DCHECK(CalledOnValidThread());
playing_ = playing;
}
float volume() const {
DCHECK(CalledOnValidThread());
return volume_;
}
void set_volume(float volume) {
DCHECK(CalledOnValidThread());
volume_ = volume;
}
private:
bool playing_;
float volume_;
};
WebRtcAudioRenderer(
const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
int source_render_view_id,
int source_render_frame_id,
int session_id,
int sample_rate,
int frames_per_buffer);
// Initialize function called by clients like WebRtcAudioDeviceImpl.
// Stop() has to be called before |source| is deleted.
bool Initialize(WebRtcAudioRendererSource* source);
// When sharing a single instance of WebRtcAudioRenderer between multiple
// users (e.g. WebMediaPlayerMS), call this method to create a proxy object
// that maintains the Play and Stop states per caller.
// The wrapper ensures that Play() won't be called when the caller's state
// is "playing", Pause() won't be called when the state already is "paused"
// etc and similarly maintains the same state for Stop().
// When Stop() is called or when the proxy goes out of scope, the proxy
// will ensure that Pause() is called followed by a call to Stop(), which
// is the usage pattern that WebRtcAudioRenderer requires.
scoped_refptr<MediaStreamAudioRenderer> CreateSharedAudioRendererProxy(
const scoped_refptr<webrtc::MediaStreamInterface>& media_stream);
// Used to DCHECK on the expected state.
bool IsStarted() const;
// Accessors to the sink audio parameters.
int channels() const { return sink_params_.channels(); }
int sample_rate() const { return sink_params_.sample_rate(); }
private:
// MediaStreamAudioRenderer implementation. This is private since we want
// callers to use proxy objects.
// TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl?
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;
// Called when an audio renderer, either the main or a proxy, starts playing.
// Here we maintain a reference count of how many renderers are currently
// playing so that the shared play state of all the streams can be reflected
// correctly.
void EnterPlayState();
// Called when an audio renderer, either the main or a proxy, is paused.
// See EnterPlayState for more details.
void EnterPauseState();
protected:
virtual ~WebRtcAudioRenderer();
private:
enum State {
UNINITIALIZED,
PLAYING,
PAUSED,
};
// Holds raw pointers to PlaingState objects. Ownership is managed outside
// of this type.
typedef std::vector<PlayingState*> PlayingStates;
// Maps an audio source to a list of playing states that collectively hold
// volume information for that source.
typedef std::map<webrtc::AudioSourceInterface*, PlayingStates>
SourcePlayingStates;
// 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);
// Goes through all renderers for the |source| and applies the proper
// volume scaling for the source based on the volume(s) of the renderer(s).
void UpdateSourceVolume(webrtc::AudioSourceInterface* source);
// Tracks a playing state. The state must be playing when this method
// is called.
// Returns true if the state was added, false if it was already being tracked.
bool AddPlayingState(webrtc::AudioSourceInterface* source,
PlayingState* state);
// Removes a playing state for an audio source.
// Returns true if the state was removed from the internal map, false if
// it had already been removed or if the source isn't being rendered.
bool RemovePlayingState(webrtc::AudioSourceInterface* source,
PlayingState* state);
// Called whenever the Play/Pause state changes of any of the renderers
// or if the volume of any of them is changed.
// Here we update the shared Play state and apply volume scaling to all audio
// sources associated with the |media_stream| based on the collective volume
// of playing renderers.
void OnPlayStateChanged(
const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
PlayingState* state);
// The render view and frame in which the audio is rendered into |sink_|.
const int source_render_view_id_;
const int source_render_frame_id_;
const int session_id_;
// The sink (destination) for rendered audio.
scoped_refptr<media::AudioOutputDevice> sink_;
// The media stream that holds the audio tracks that this renderer renders.
const scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
// Audio data source from the browser process.
WebRtcAudioRendererSource* source_;
// Protects access to |state_|, |source_| and |sink_|.
base::Lock lock_;
// Ref count for the MediaPlayers which are playing audio.
int play_ref_count_;
// Ref count for the MediaPlayers which have called Start() but not Stop().
int start_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_;
// Delay due to the FIFO in milliseconds.
int fifo_delay_milliseconds_;
// Saved volume and playing state of the root renderer.
PlayingState playing_state_;
// Audio params used by the sink of the renderer.
media::AudioParameters sink_params_;
// Maps audio sources to a list of active audio renderers.
// Pointers to PlayingState objects are only kept in this map while the
// associated renderer is actually playing the stream. Ownership of the
// state objects lies with the renderers and they must leave the playing state
// before being destructed (PlayingState object goes out of scope).
SourcePlayingStates source_playing_states_;
DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
};
} // namespace content
#endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_