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_LOCAL_AUDIO_RENDERER_H_ 6 #define CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ 7 8 #include <vector> 9 10 #include "base/callback.h" 11 #include "base/memory/ref_counted.h" 12 #include "base/synchronization/lock.h" 13 #include "base/threading/thread_checker.h" 14 #include "content/common/content_export.h" 15 #include "content/renderer/media/media_stream_audio_renderer.h" 16 #include "content/renderer/media/webrtc_audio_device_impl.h" 17 #include "content/renderer/media/webrtc_local_audio_track.h" 18 19 namespace media { 20 class AudioBus; 21 class AudioFifo; 22 class AudioOutputDevice; 23 class AudioParameters; 24 } 25 26 namespace content { 27 28 class WebRtcAudioCapturer; 29 30 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering 31 // local audio media stream tracks, 32 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack 33 // It also implements media::AudioRendererSink::RenderCallback to render audio 34 // data provided from a WebRtcLocalAudioTrack source. 35 // When the audio layer in the browser process asks for data to render, this 36 // class provides the data by implementing the WebRtcAudioCapturerSink 37 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective. 38 // TODO(henrika): improve by using similar principles as in RTCVideoRenderer 39 // which register itself to the video track when the provider is started and 40 // deregisters itself when it is stopped. 41 // Tracking this at http://crbug.com/164813. 42 class CONTENT_EXPORT WebRtcLocalAudioRenderer 43 : NON_EXPORTED_BASE(public MediaStreamAudioRenderer), 44 NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback), 45 NON_EXPORTED_BASE(public WebRtcAudioCapturerSink) { 46 public: 47 // Creates a local renderer and registers a capturing |source| object. 48 // The |source| is owned by the WebRtcAudioDeviceImpl. 49 // Called on the main thread. 50 WebRtcLocalAudioRenderer(WebRtcLocalAudioTrack* audio_track, 51 int source_render_view_id); 52 53 // MediaStreamAudioRenderer implementation. 54 // Called on the main thread. 55 virtual void Start() OVERRIDE; 56 virtual void Stop() OVERRIDE; 57 virtual void Play() OVERRIDE; 58 virtual void Pause() OVERRIDE; 59 virtual void SetVolume(float volume) OVERRIDE; 60 virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE; 61 virtual bool IsLocalRenderer() const OVERRIDE; 62 63 const base::TimeDelta& total_render_time() const { 64 return total_render_time_; 65 } 66 67 protected: 68 virtual ~WebRtcLocalAudioRenderer(); 69 70 private: 71 // WebRtcAudioCapturerSink implementation. 72 73 // Called on the AudioInputDevice worker thread. 74 virtual int CaptureData(const std::vector<int>& channels, 75 const int16* audio_data, 76 int sample_rate, 77 int number_of_channels, 78 int number_of_frames, 79 int audio_delay_milliseconds, 80 int current_volume, 81 bool need_audio_processing) OVERRIDE; 82 83 // Can be called on different user thread. 84 virtual void SetCaptureFormat(const media::AudioParameters& params) OVERRIDE; 85 86 // media::AudioRendererSink::RenderCallback implementation. 87 // Render() is called on the AudioOutputDevice thread and OnRenderError() 88 // on the IO thread. 89 virtual int Render(media::AudioBus* audio_bus, 90 int audio_delay_milliseconds) OVERRIDE; 91 virtual void OnRenderError() OVERRIDE; 92 93 // The audio track which provides data to render. Given that this class 94 // implements local loopback, the audio track is getting data from a capture 95 // instance like a selected microphone and forwards the recorded data to its 96 // sinks. The recorded data is stored in a FIFO and consumed 97 // by this class when the sink asks for new data. 98 // The WebRtcAudioCapturer is today created by WebRtcAudioDeviceImpl. 99 scoped_refptr<WebRtcLocalAudioTrack> audio_track_; 100 101 // The render view in which the audio is rendered into |sink_|. 102 const int source_render_view_id_; 103 104 // The sink (destination) for rendered audio. 105 scoped_refptr<media::AudioOutputDevice> sink_; 106 107 // Used to DCHECK that we are called on the correct thread. 108 base::ThreadChecker thread_checker_; 109 110 // Contains copies of captured audio frames. 111 scoped_ptr<media::AudioFifo> loopback_fifo_; 112 113 // Stores last time a render callback was received. The time difference 114 // between a new time stamp and this value can be used to derive the 115 // total render time. 116 base::Time last_render_time_; 117 118 // Keeps track of total time audio has been rendered. 119 base::TimeDelta total_render_time_; 120 121 // The audio parameters used by the renderer. 122 media::AudioParameters audio_params_; 123 124 // Set when playing, cleared when paused. 125 bool playing_; 126 127 // Protects |loopback_fifo_|, |playing_| and |sink_|. 128 mutable base::Lock thread_lock_; 129 130 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); 131 }; 132 133 } // namespace content 134 135 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ 136