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/message_loop/message_loop_proxy.h" 13 #include "base/synchronization/lock.h" 14 #include "base/threading/thread_checker.h" 15 #include "content/common/content_export.h" 16 #include "content/public/renderer/media_stream_audio_sink.h" 17 #include "content/renderer/media/media_stream_audio_renderer.h" 18 #include "content/renderer/media/webrtc_audio_device_impl.h" 19 #include "content/renderer/media/webrtc_local_audio_track.h" 20 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" 21 22 namespace media { 23 class AudioBus; 24 class AudioBlockFifo; 25 class AudioOutputDevice; 26 class AudioParameters; 27 } 28 29 namespace content { 30 31 class WebRtcAudioCapturer; 32 33 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering 34 // local audio media stream tracks, 35 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack 36 // It also implements media::AudioRendererSink::RenderCallback to render audio 37 // data provided from a WebRtcLocalAudioTrack source. 38 // When the audio layer in the browser process asks for data to render, this 39 // class provides the data by implementing the MediaStreamAudioSink 40 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective. 41 // TODO(henrika): improve by using similar principles as in RTCVideoRenderer 42 // which register itself to the video track when the provider is started and 43 // deregisters itself when it is stopped. 44 // Tracking this at http://crbug.com/164813. 45 class CONTENT_EXPORT WebRtcLocalAudioRenderer NON_EXPORTED_BASE(public MediaStreamAudioRenderer)46 : NON_EXPORTED_BASE(public MediaStreamAudioRenderer), 47 NON_EXPORTED_BASE(public MediaStreamAudioSink), 48 NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback) { 49 public: 50 // Creates a local renderer and registers a capturing |source| object. 51 // The |source| is owned by the WebRtcAudioDeviceImpl. 52 // Called on the main thread. 53 WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track, 54 int source_render_view_id, 55 int source_render_frame_id, 56 int session_id, 57 int frames_per_buffer); 58 59 // MediaStreamAudioRenderer implementation. 60 // Called on the main thread. 61 virtual void Start() OVERRIDE; 62 virtual void Stop() OVERRIDE; 63 virtual void Play() OVERRIDE; 64 virtual void Pause() OVERRIDE; 65 virtual void SetVolume(float volume) OVERRIDE; 66 virtual base::TimeDelta GetCurrentRenderTime() const OVERRIDE; 67 virtual bool IsLocalRenderer() const OVERRIDE; 68 69 const base::TimeDelta& total_render_time() const { 70 return total_render_time_; 71 } 72 73 protected: 74 virtual ~WebRtcLocalAudioRenderer(); 75 76 private: 77 // MediaStreamAudioSink implementation. 78 79 // Called on the AudioInputDevice worker thread. 80 virtual void OnData(const int16* audio_data, 81 int sample_rate, 82 int number_of_channels, 83 int number_of_frames) OVERRIDE; 84 85 // Called on the AudioInputDevice worker thread. 86 virtual void OnSetFormat(const media::AudioParameters& params) OVERRIDE; 87 88 // media::AudioRendererSink::RenderCallback implementation. 89 // Render() is called on the AudioOutputDevice thread and OnRenderError() 90 // on the IO thread. 91 virtual int Render(media::AudioBus* audio_bus, 92 int audio_delay_milliseconds) OVERRIDE; 93 virtual void OnRenderError() OVERRIDE; 94 95 // Initializes and starts the |sink_| if 96 // we have received valid |source_params_| && 97 // |playing_| has been set to true && 98 // |volume_| is not zero. 99 void MaybeStartSink(); 100 101 // Sets new |source_params_| and then re-initializes and restarts |sink_|. 102 void ReconfigureSink(const media::AudioParameters& params); 103 104 // The audio track which provides data to render. Given that this class 105 // implements local loopback, the audio track is getting data from a capture 106 // instance like a selected microphone and forwards the recorded data to its 107 // sinks. The recorded data is stored in a FIFO and consumed 108 // by this class when the sink asks for new data. 109 // This class is calling MediaStreamAudioSink::AddToAudioTrack() and 110 // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect 111 // with the audio track. 112 blink::WebMediaStreamTrack audio_track_; 113 114 // The render view and frame in which the audio is rendered into |sink_|. 115 const int source_render_view_id_; 116 const int source_render_frame_id_; 117 const int session_id_; 118 119 // MessageLoop associated with the single thread that performs all control 120 // tasks. Set to the MessageLoop that invoked the ctor. 121 const scoped_refptr<base::MessageLoopProxy> message_loop_; 122 123 // The sink (destination) for rendered audio. 124 scoped_refptr<media::AudioOutputDevice> sink_; 125 126 // Contains copies of captured audio frames. 127 scoped_ptr<media::AudioBlockFifo> loopback_fifo_; 128 129 // Stores last time a render callback was received. The time difference 130 // between a new time stamp and this value can be used to derive the 131 // total render time. 132 base::TimeTicks last_render_time_; 133 134 // Keeps track of total time audio has been rendered. 135 base::TimeDelta total_render_time_; 136 137 // The audio parameters of the capture source. 138 // Must only be touched on the main thread. 139 media::AudioParameters source_params_; 140 141 // The audio parameters used by the sink. 142 // Must only be touched on the main thread. 143 media::AudioParameters sink_params_; 144 145 // Set when playing, cleared when paused. 146 bool playing_; 147 148 // Protects |loopback_fifo_|, |playing_| and |sink_|. 149 mutable base::Lock thread_lock_; 150 151 // The preferred buffer size provided via the ctor. 152 const int frames_per_buffer_; 153 154 // The preferred device id of the output device or empty for the default 155 // output device. 156 const std::string output_device_id_; 157 158 // Cache value for the volume. 159 float volume_; 160 161 // Flag to indicate whether |sink_| has been started yet. 162 bool sink_started_; 163 164 // Used to DCHECK that some methods are called on the capture audio thread. 165 base::ThreadChecker capture_thread_checker_; 166 167 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); 168 }; 169 170 } // namespace content 171 172 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ 173