| // 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(); } |
| int frames_per_buffer() const { return sink_params_.frames_per_buffer(); } |
| |
| 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_|, |sink_| and |current_time_|. |
| mutable 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_; |
| |
| base::TimeDelta current_time_; |
| |
| // 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_; |
| |
| // Used for triggering new UMA histogram. Counts number of render |
| // callbacks modulo |kNumCallbacksBetweenRenderTimeHistograms|. |
| int render_callback_count_; |
| |
| DISALLOW_IMPLICIT_CONSTRUCTORS(WebRtcAudioRenderer); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_WEBRTC_AUDIO_RENDERER_H_ |