| // 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/synchronization/lock.h" |
| #include "base/threading/thread_checker.h" |
| #include "content/common/content_export.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" |
| |
| namespace media { |
| class AudioBus; |
| class AudioFifo; |
| 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 WebRtcAudioCapturerSink |
| // 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 media::AudioRendererSink::RenderCallback), |
| NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) { |
| public: |
| // Creates a local renderer and registers a capturing |source| object. |
| // The |source| is owned by the WebRtcAudioDeviceImpl. |
| // Called on the main thread. |
| WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track, |
| int source_render_view_id); |
| |
| // MediaStreamAudioRenderer implementation. |
| // Called on the main thread. |
| virtual void Start() OVERRIDE; |
| virtual void Stop() OVERRIDE; |
| virtual void Play() OVERRIDE; |
| virtual void Pause() OVERRIDE; |
| virtual void SetVolume(float volume) OVERRIDE; |
| virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE; |
| virtual bool IsLocalRenderer() const OVERRIDE; |
| |
| const base::TimeDelta& total_render_time() const { |
| return total_render_time_; |
| } |
| |
| protected: |
| virtual ~WebRtcLocalAudioRenderer(); |
| |
| private: |
| // WebRtcAudioCapturerSink implementation. |
| |
| // Called on the AudioInputDevice worker thread. |
| virtual int CaptureData(const std::vector<int>& channels, |
| const int16* audio_data, |
| int sample_rate, |
| int number_of_channels, |
| int number_of_frames, |
| int audio_delay_milliseconds, |
| int current_volume, |
| bool need_audio_processing) OVERRIDE; |
| |
| // Can be called on different user thread. |
| virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; |
| |
| // media::AudioRendererSink::RenderCallback implementation. |
| // Render() is called on the AudioOutputDevice thread and OnRenderError() |
| // on the IO thread. |
| virtual int Render(media::AudioBus* audio_bus, |
| int audio_delay_milliseconds) OVERRIDE; |
| virtual void OnRenderError() OVERRIDE; |
| |
| // 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. |
| // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl. |
| scoped_refptr<WebRtcLocalAudioTrack> audio_track_; |
| |
| // The render view in which the audio is rendered into |sink_|. |
| const int source_render_view_id_; |
| |
| // The sink (destination) for rendered audio. |
| scoped_refptr<media::AudioOutputDevice> sink_; |
| |
| // Used to DCHECK that we are called on the correct thread. |
| base::ThreadChecker thread_checker_; |
| |
| // Contains copies of captured audio frames. |
| scoped_ptr<media::AudioFifo> loopback_fifo_; |
| |
| // 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::Time last_render_time_; |
| |
| // Keeps track of total time audio has been rendered. |
| base::TimeDelta total_render_time_; |
| |
| // The audio parameters used by the renderer. |
| media::AudioParameters audio_params_; |
| |
| // Set when playing, cleared when paused. |
| bool playing_; |
| |
| // Protects |loopback_fifo_|, |playing_| and |sink_|. |
| mutable base::Lock thread_lock_; |
| |
| DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); |
| }; |
| |
| } // namespace content |
| |
| #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ |