| // 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_LOCAL_AUDIO_RENDERER_H_ |
| #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ |
| |
| #include <vector> |
| |
| #include "base/callback.h" |
| #include "base/memory/ref_counted.h" |
| #include "base/message_loop/message_loop_proxy.h" |
| #include "base/synchronization/lock.h" |
| #include "base/threading/thread_checker.h" |
| #include "content/common/content_export.h" |
| #include "content/public/renderer/media_stream_audio_sink.h" |
| #include "content/renderer/media/media_stream_audio_renderer.h" |
| #include "content/renderer/media/webrtc_audio_device_impl.h" |
| #include "content/renderer/media/webrtc_local_audio_track.h" |
| #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" |
| |
| namespace media { |
| class AudioBus; |
| class AudioShifter; |
| class AudioOutputDevice; |
| class AudioParameters; |
| } |
| |
| namespace content { |
| |
| class WebRtcAudioCapturer; |
| |
| // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering |
| // local audio media stream tracks, |
| // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack |
| // It also implements media::AudioRendererSink::RenderCallback to render audio |
| // data provided from a WebRtcLocalAudioTrack source. |
| // When the audio layer in the browser process asks for data to render, this |
| // class provides the data by implementing the MediaStreamAudioSink |
| // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective. |
| // TODO(henrika): improve by using similar principles as in RTCVideoRenderer |
| // which register itself to the video track when the provider is started and |
| // deregisters itself when it is stopped. |
| // Tracking this at http://crbug.com/164813. |
| class CONTENT_EXPORT WebRtcLocalAudioRenderer |
| : NON_EXPORTED_BASE(public MediaStreamAudioRenderer), |
| NON_EXPORTED_BASE(public MediaStreamAudioSink), |
| NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback) { |
| public: |
| // Creates a local renderer and registers a capturing |source| object. |
| // The |source| is owned by the WebRtcAudioDeviceImpl. |
| // Called on the main thread. |
| WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track, |
| int source_render_frame_id, |
| int session_id, |
| int frames_per_buffer); |
| |
| // MediaStreamAudioRenderer implementation. |
| // Called on the main thread. |
| void Start() override; |
| void Stop() override; |
| void Play() override; |
| void Pause() override; |
| void SetVolume(float volume) override; |
| base::TimeDelta GetCurrentRenderTime() const override; |
| bool IsLocalRenderer() const override; |
| |
| const base::TimeDelta& total_render_time() const { |
| return total_render_time_; |
| } |
| |
| protected: |
| ~WebRtcLocalAudioRenderer() override; |
| |
| private: |
| // MediaStreamAudioSink implementation. |
| |
| // Called on the AudioInputDevice worker thread. |
| void OnData(const media::AudioBus& audio_bus, |
| base::TimeTicks estimated_capture_time) override; |
| |
| // Called on the AudioInputDevice worker thread. |
| void OnSetFormat(const media::AudioParameters& params) override; |
| |
| // media::AudioRendererSink::RenderCallback implementation. |
| // Render() is called on the AudioOutputDevice thread and OnRenderError() |
| // on the IO thread. |
| int Render(media::AudioBus* audio_bus, int audio_delay_milliseconds) override; |
| void OnRenderError() override; |
| |
| // Initializes and starts the |sink_| if |
| // we have received valid |source_params_| && |
| // |playing_| has been set to true && |
| // |volume_| is not zero. |
| void MaybeStartSink(); |
| |
| // Sets new |source_params_| and then re-initializes and restarts |sink_|. |
| void ReconfigureSink(const media::AudioParameters& params); |
| |
| // The audio track which provides data to render. Given that this class |
| // implements local loopback, the audio track is getting data from a capture |
| // instance like a selected microphone and forwards the recorded data to its |
| // sinks. The recorded data is stored in a FIFO and consumed |
| // by this class when the sink asks for new data. |
| // This class is calling MediaStreamAudioSink::AddToAudioTrack() and |
| // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect |
| // with the audio track. |
| blink::WebMediaStreamTrack audio_track_; |
| |
| // The render view and frame in which the audio is rendered into |sink_|. |
| const int source_render_frame_id_; |
| const int session_id_; |
| |
| // MessageLoop associated with the single thread that performs all control |
| // tasks. Set to the MessageLoop that invoked the ctor. |
| const scoped_refptr<base::MessageLoopProxy> message_loop_; |
| |
| // The sink (destination) for rendered audio. |
| scoped_refptr<media::AudioOutputDevice> sink_; |
| |
| // This does all the synchronization/resampling/smoothing. |
| scoped_ptr<media::AudioShifter> audio_shifter_; |
| |
| // Stores last time a render callback was received. The time difference |
| // between a new time stamp and this value can be used to derive the |
| // total render time. |
| base::TimeTicks last_render_time_; |
| |
| // Keeps track of total time audio has been rendered. |
| base::TimeDelta total_render_time_; |
| |
| // The audio parameters of the capture source. |
| // Must only be touched on the main thread. |
| media::AudioParameters source_params_; |
| |
| // The audio parameters used by the sink. |
| // Must only be touched on the main thread. |
| media::AudioParameters sink_params_; |
| |
| // Set when playing, cleared when paused. |
| bool playing_; |
| |
| // Protects |audio_shifter_|, |playing_| and |sink_|. |
| mutable base::Lock thread_lock_; |
| |
| // The preferred buffer size provided via the ctor. |
| const int frames_per_buffer_; |
| |
| // The preferred device id of the output device or empty for the default |
| // output device. |
| const std::string output_device_id_; |
| |
| // Cache value for the volume. |
| float volume_; |
| |
| // Flag to indicate whether |sink_| has been started yet. |
| bool sink_started_; |
| |
| // Used to DCHECK that some methods are called on the capture audio thread. |
| base::ThreadChecker capture_thread_checker_; |
| |
| DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ |