1// Copyright (c) 2012 The Chromium Authors. All rights reserved.
2// Use of this source code is governed by a BSD-style license that can be
3// found in the LICENSE file.
4
5#ifndef CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
6#define CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
7
8#include "base/memory/ref_counted.h"
9#include "base/synchronization/lock.h"
10#include "base/threading/non_thread_safe.h"
11#include "base/threading/thread_checker.h"
12#include "content/renderer/media/media_stream_audio_renderer.h"
13#include "content/renderer/media/webrtc_audio_device_impl.h"
14#include "media/base/audio_decoder.h"
15#include "media/base/audio_pull_fifo.h"
16#include "media/base/audio_renderer_sink.h"
17#include "media/base/channel_layout.h"
18
19namespace media {
20class AudioOutputDevice;
21}  // namespace media
22
23namespace webrtc {
24class AudioSourceInterface;
25class MediaStreamInterface;
26}  // namespace webrtc
27
28namespace content {
29
30class WebRtcAudioRendererSource;
31
32// This renderer handles calls from the pipeline and WebRtc ADM. It is used
33// for connecting WebRtc MediaStream with the audio pipeline.
34class CONTENT_EXPORT WebRtcAudioRenderer
35    : NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback),
36      NON_EXPORTED_BASE(public MediaStreamAudioRenderer) {
37 public:
38  // This is a little utility class that holds the configured state of an audio
39  // stream.
40  // It is used by both WebRtcAudioRenderer and SharedAudioRenderer (see cc
41  // file) so a part of why it exists is to avoid code duplication and track
42  // the state in the same way in WebRtcAudioRenderer and SharedAudioRenderer.
43  class PlayingState : public base::NonThreadSafe {
44   public:
45    PlayingState() : playing_(false), volume_(1.0f) {}
46
47    bool playing() const {
48      DCHECK(CalledOnValidThread());
49      return playing_;
50    }
51
52    void set_playing(bool playing) {
53      DCHECK(CalledOnValidThread());
54      playing_ = playing;
55    }
56
57    float volume() const {
58      DCHECK(CalledOnValidThread());
59      return volume_;
60    }
61
62    void set_volume(float volume) {
63      DCHECK(CalledOnValidThread());
64      volume_ = volume;
65    }
66
67   private:
68    bool playing_;
69    float volume_;
70  };
71
72  WebRtcAudioRenderer(
73      const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
74      int source_render_view_id,
75      int source_render_frame_id,
76      int session_id,
77      int sample_rate,
78      int frames_per_buffer);
79
80  // Initialize function called by clients like WebRtcAudioDeviceImpl.
81  // Stop() has to be called before |source| is deleted.
82  bool Initialize(WebRtcAudioRendererSource* source);
83
84  // When sharing a single instance of WebRtcAudioRenderer between multiple
85  // users (e.g. WebMediaPlayerMS), call this method to create a proxy object
86  // that maintains the Play and Stop states per caller.
87  // The wrapper ensures that Play() won't be called when the caller's state
88  // is "playing", Pause() won't be called when the state already is "paused"
89  // etc and similarly maintains the same state for Stop().
90  // When Stop() is called or when the proxy goes out of scope, the proxy
91  // will ensure that Pause() is called followed by a call to Stop(), which
92  // is the usage pattern that WebRtcAudioRenderer requires.
93  scoped_refptr<MediaStreamAudioRenderer> CreateSharedAudioRendererProxy(
94      const scoped_refptr<webrtc::MediaStreamInterface>& media_stream);
95
96  // Used to DCHECK on the expected state.
97  bool IsStarted() const;
98
99  // Accessors to the sink audio parameters.
100  int channels() const { return sink_params_.channels(); }
101  int sample_rate() const { return sink_params_.sample_rate(); }
102
103 private:
104  // MediaStreamAudioRenderer implementation.  This is private since we want
105  // callers to use proxy objects.
106  // TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl?
107  virtual void Start() OVERRIDE;
108  virtual void Play() OVERRIDE;
109  virtual void Pause() OVERRIDE;
110  virtual void Stop() OVERRIDE;
111  virtual void SetVolume(float volume) OVERRIDE;
112  virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE;
113  virtual bool IsLocalRenderer() const OVERRIDE;
114
115  // Called when an audio renderer, either the main or a proxy, starts playing.
116  // Here we maintain a reference count of how many renderers are currently
117  // playing so that the shared play state of all the streams can be reflected
118  // correctly.
119  void EnterPlayState();
120
121  // Called when an audio renderer, either the main or a proxy, is paused.
122  // See EnterPlayState for more details.
123  void EnterPauseState();
124
125 protected:
126  virtual ~WebRtcAudioRenderer();
127
128 private:
129  enum State {
130    UNINITIALIZED,
131    PLAYING,
132    PAUSED,
133  };
134
135  // Holds raw pointers to PlaingState objects.  Ownership is managed outside
136  // of this type.
137  typedef std::vector<PlayingState*> PlayingStates;
138  // Maps an audio source to a list of playing states that collectively hold
139  // volume information for that source.
140  typedef std::map<webrtc::AudioSourceInterface*, PlayingStates>
141      SourcePlayingStates;
142
143  // Used to DCHECK that we are called on the correct thread.
144  base::ThreadChecker thread_checker_;
145
146  // Flag to keep track the state of the renderer.
147  State state_;
148
149  // media::AudioRendererSink::RenderCallback implementation.
150  // These two methods are called on the AudioOutputDevice worker thread.
151  virtual int Render(media::AudioBus* audio_bus,
152                     int audio_delay_milliseconds) OVERRIDE;
153  virtual void OnRenderError() OVERRIDE;
154
155  // Called by AudioPullFifo when more data is necessary.
156  // This method is called on the AudioOutputDevice worker thread.
157  void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus);
158
159  // Goes through all renderers for the |source| and applies the proper
160  // volume scaling for the source based on the volume(s) of the renderer(s).
161  void UpdateSourceVolume(webrtc::AudioSourceInterface* source);
162
163  // Tracks a playing state.  The state must be playing when this method
164  // is called.
165  // Returns true if the state was added, false if it was already being tracked.
166  bool AddPlayingState(webrtc::AudioSourceInterface* source,
167                       PlayingState* state);
168  // Removes a playing state for an audio source.
169  // Returns true if the state was removed from the internal map, false if
170  // it had already been removed or if the source isn't being rendered.
171  bool RemovePlayingState(webrtc::AudioSourceInterface* source,
172                          PlayingState* state);
173
174  // Called whenever the Play/Pause state changes of any of the renderers
175  // or if the volume of any of them is changed.
176  // Here we update the shared Play state and apply volume scaling to all audio
177  // sources associated with the |media_stream| based on the collective volume
178  // of playing renderers.
179  void OnPlayStateChanged(
180      const scoped_refptr<webrtc::MediaStreamInterface>& media_stream,
181      PlayingState* state);
182
183  // The render view and frame in which the audio is rendered into |sink_|.
184  const int source_render_view_id_;
185  const int source_render_frame_id_;
186  const int session_id_;
187
188  // The sink (destination) for rendered audio.
189  scoped_refptr<media::AudioOutputDevice> sink_;
190
191  // The media stream that holds the audio tracks that this renderer renders.
192  const scoped_refptr<webrtc::MediaStreamInterface> media_stream_;
193
194  // Audio data source from the browser process.
195  WebRtcAudioRendererSource* source_;
196
197  // Protects access to |state_|, |source_|, |sink_| and |current_time_|.
198  mutable base::Lock lock_;
199
200  // Ref count for the MediaPlayers which are playing audio.
201  int play_ref_count_;
202
203  // Ref count for the MediaPlayers which have called Start() but not Stop().
204  int start_ref_count_;
205
206  // Used to buffer data between the client and the output device in cases where
207  // the client buffer size is not the same as the output device buffer size.
208  scoped_ptr<media::AudioPullFifo> audio_fifo_;
209
210  // Contains the accumulated delay estimate which is provided to the WebRTC
211  // AEC.
212  int audio_delay_milliseconds_;
213
214  // Delay due to the FIFO in milliseconds.
215  int fifo_delay_milliseconds_;
216
217  base::TimeDelta current_time_;
218
219  // Saved volume and playing state of the root renderer.
220  PlayingState playing_state_;
221
222  // Audio params used by the sink of the renderer.
223  media::AudioParameters sink_params_;
224
225  // Maps audio sources to a list of active audio renderers.
226  // Pointers to PlayingState objects are only kept in this map while the
227  // associated renderer is actually playing the stream.  Ownership of the
228  // state objects lies with the renderers and they must leave the playing state
229  // before being destructed (PlayingState object goes out of scope).
230  SourcePlayingStates source_playing_states_;
231
232  DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer);
233};
234
235}  // namespace content
236
237#endif  // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_
238