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 19 namespace media { 20 class AudioOutputDevice; 21 } // namespace media 22 23 namespace webrtc { 24 class AudioSourceInterface; 25 class MediaStreamInterface; 26 } // namespace webrtc 27 28 namespace content { 29 30 class 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. 34 class 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 int frames_per_buffer() const { return sink_params_.frames_per_buffer(); } 103 104 private: 105 // MediaStreamAudioRenderer implementation. This is private since we want 106 // callers to use proxy objects. 107 // TODO(tommi): Make the MediaStreamAudioRenderer implementation a pimpl? 108 virtual void Start() OVERRIDE; 109 virtual void Play() OVERRIDE; 110 virtual void Pause() OVERRIDE; 111 virtual void Stop() OVERRIDE; 112 virtual void SetVolume(float volume) OVERRIDE; 113 virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE; 114 virtual bool IsLocalRenderer() const OVERRIDE; 115 116 // Called when an audio renderer, either the main or a proxy, starts playing. 117 // Here we maintain a reference count of how many renderers are currently 118 // playing so that the shared play state of all the streams can be reflected 119 // correctly. 120 void EnterPlayState(); 121 122 // Called when an audio renderer, either the main or a proxy, is paused. 123 // See EnterPlayState for more details. 124 void EnterPauseState(); 125 126 protected: 127 virtual ~WebRtcAudioRenderer(); 128 129 private: 130 enum State { 131 UNINITIALIZED, 132 PLAYING, 133 PAUSED, 134 }; 135 136 // Holds raw pointers to PlaingState objects. Ownership is managed outside 137 // of this type. 138 typedef std::vector<PlayingState*> PlayingStates; 139 // Maps an audio source to a list of playing states that collectively hold 140 // volume information for that source. 141 typedef std::map<webrtc::AudioSourceInterface*, PlayingStates> 142 SourcePlayingStates; 143 144 // Used to DCHECK that we are called on the correct thread. 145 base::ThreadChecker thread_checker_; 146 147 // Flag to keep track the state of the renderer. 148 State state_; 149 150 // media::AudioRendererSink::RenderCallback implementation. 151 // These two methods are called on the AudioOutputDevice worker thread. 152 virtual int Render(media::AudioBus* audio_bus, 153 int audio_delay_milliseconds) OVERRIDE; 154 virtual void OnRenderError() OVERRIDE; 155 156 // Called by AudioPullFifo when more data is necessary. 157 // This method is called on the AudioOutputDevice worker thread. 158 void SourceCallback(int fifo_frame_delay, media::AudioBus* audio_bus); 159 160 // Goes through all renderers for the |source| and applies the proper 161 // volume scaling for the source based on the volume(s) of the renderer(s). 162 void UpdateSourceVolume(webrtc::AudioSourceInterface* source); 163 164 // Tracks a playing state. The state must be playing when this method 165 // is called. 166 // Returns true if the state was added, false if it was already being tracked. 167 bool AddPlayingState(webrtc::AudioSourceInterface* source, 168 PlayingState* state); 169 // Removes a playing state for an audio source. 170 // Returns true if the state was removed from the internal map, false if 171 // it had already been removed or if the source isn't being rendered. 172 bool RemovePlayingState(webrtc::AudioSourceInterface* source, 173 PlayingState* state); 174 175 // Called whenever the Play/Pause state changes of any of the renderers 176 // or if the volume of any of them is changed. 177 // Here we update the shared Play state and apply volume scaling to all audio 178 // sources associated with the |media_stream| based on the collective volume 179 // of playing renderers. 180 void OnPlayStateChanged( 181 const scoped_refptr<webrtc::MediaStreamInterface>& media_stream, 182 PlayingState* state); 183 184 // The render view and frame in which the audio is rendered into |sink_|. 185 const int source_render_view_id_; 186 const int source_render_frame_id_; 187 const int session_id_; 188 189 // The sink (destination) for rendered audio. 190 scoped_refptr<media::AudioOutputDevice> sink_; 191 192 // The media stream that holds the audio tracks that this renderer renders. 193 const scoped_refptr<webrtc::MediaStreamInterface> media_stream_; 194 195 // Audio data source from the browser process. 196 WebRtcAudioRendererSource* source_; 197 198 // Protects access to |state_|, |source_|, |sink_| and |current_time_|. 199 mutable base::Lock lock_; 200 201 // Ref count for the MediaPlayers which are playing audio. 202 int play_ref_count_; 203 204 // Ref count for the MediaPlayers which have called Start() but not Stop(). 205 int start_ref_count_; 206 207 // Used to buffer data between the client and the output device in cases where 208 // the client buffer size is not the same as the output device buffer size. 209 scoped_ptr<media::AudioPullFifo> audio_fifo_; 210 211 // Contains the accumulated delay estimate which is provided to the WebRTC 212 // AEC. 213 int audio_delay_milliseconds_; 214 215 // Delay due to the FIFO in milliseconds. 216 int fifo_delay_milliseconds_; 217 218 base::TimeDelta current_time_; 219 220 // Saved volume and playing state of the root renderer. 221 PlayingState playing_state_; 222 223 // Audio params used by the sink of the renderer. 224 media::AudioParameters sink_params_; 225 226 // Maps audio sources to a list of active audio renderers. 227 // Pointers to PlayingState objects are only kept in this map while the 228 // associated renderer is actually playing the stream. Ownership of the 229 // state objects lies with the renderers and they must leave the playing state 230 // before being destructed (PlayingState object goes out of scope). 231 SourcePlayingStates source_playing_states_; 232 233 // Used for triggering new UMA histogram. Counts number of render 234 // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|. 235 int render_callback_count_; 236 237 DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer); 238 }; 239 240 } // namespace content 241 242 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ 243