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| 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 <stdint.h> | |
| 9 | |
| 10 #include <string> | |
| 11 #include <vector> | |
| 12 | |
| 13 #include "base/callback.h" | |
| 14 #include "base/macros.h" | |
| 15 #include "base/memory/ref_counted.h" | |
| 16 #include "base/single_thread_task_runner.h" | |
| 17 #include "base/synchronization/lock.h" | |
| 18 #include "base/threading/thread_checker.h" | |
| 19 #include "content/common/content_export.h" | |
| 20 #include "content/public/renderer/media_stream_audio_renderer.h" | |
| 21 #include "content/public/renderer/media_stream_audio_sink.h" | |
| 22 #include "content/renderer/media/webrtc_audio_device_impl.h" | |
| 23 #include "content/renderer/media/webrtc_local_audio_track.h" | |
| 24 #include "media/base/output_device.h" | |
| 25 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" | |
| 26 | |
| 27 namespace media { | |
| 28 class AudioBus; | |
| 29 class AudioShifter; | |
| 30 class AudioOutputDevice; | |
| 31 class AudioParameters; | |
| 32 } | |
| 33 | |
| 34 namespace content { | |
| 35 | |
| 36 class WebRtcAudioCapturer; | |
| 37 | |
| 38 // WebRtcLocalAudioRenderer is a MediaStreamAudioRenderer designed for rendering | |
| 39 // local audio media stream tracks, | |
| 40 // http://dev.w3.org/2011/webrtc/editor/getusermedia.html#mediastreamtrack | |
| 41 // It also implements media::AudioRendererSink::RenderCallback to render audio | |
| 42 // data provided from a WebRtcLocalAudioTrack source. | |
| 43 // When the audio layer in the browser process asks for data to render, this | |
| 44 // class provides the data by implementing the MediaStreamAudioSink | |
| 45 // interface, i.e., we are a sink seen from the WebRtcAudioCapturer perspective. | |
| 46 // TODO(henrika): improve by using similar principles as in | |
| 47 // MediaStreamVideoRendererSink which register itself to the video track when | |
| 48 // the provider is started and deregisters itself when it is stopped. Tracking | |
| 49 // this at http://crbug.com/164813. | |
| 50 class CONTENT_EXPORT WebRtcLocalAudioRenderer | |
| 51 : NON_EXPORTED_BASE(public MediaStreamAudioRenderer), | |
| 52 NON_EXPORTED_BASE(public MediaStreamAudioSink), | |
| 53 NON_EXPORTED_BASE(public media::AudioRendererSink::RenderCallback), | |
| 54 NON_EXPORTED_BASE(public media::OutputDevice) { | |
| 55 public: | |
| 56 // Creates a local renderer and registers a capturing |source| object. | |
| 57 // The |source| is owned by the WebRtcAudioDeviceImpl. | |
| 58 // Called on the main thread. | |
| 59 WebRtcLocalAudioRenderer(const blink::WebMediaStreamTrack& audio_track, | |
| 60 int source_render_frame_id, | |
| 61 int session_id, | |
| 62 const std::string& device_id, | |
| 63 const url::Origin& security_origin); | |
| 64 | |
| 65 // MediaStreamAudioRenderer implementation. | |
| 66 // Called on the main thread. | |
| 67 void Start() override; | |
| 68 void Stop() override; | |
| 69 void Play() override; | |
| 70 void Pause() override; | |
| 71 void SetVolume(float volume) override; | |
| 72 media::OutputDevice* GetOutputDevice() override; | |
| 73 base::TimeDelta GetCurrentRenderTime() const override; | |
| 74 bool IsLocalRenderer() const override; | |
| 75 | |
| 76 // media::OutputDevice implementation | |
| 77 void SwitchOutputDevice(const std::string& device_id, | |
| 78 const url::Origin& security_origin, | |
| 79 const media::SwitchOutputDeviceCB& callback) override; | |
| 80 media::AudioParameters GetOutputParameters() override; | |
| 81 media::OutputDeviceStatus GetDeviceStatus() override; | |
| 82 | |
| 83 const base::TimeDelta& total_render_time() const { | |
| 84 return total_render_time_; | |
| 85 } | |
| 86 | |
| 87 protected: | |
| 88 ~WebRtcLocalAudioRenderer() override; | |
| 89 | |
| 90 private: | |
| 91 // MediaStreamAudioSink implementation. | |
| 92 | |
| 93 // Called on the AudioInputDevice worker thread. | |
| 94 void OnData(const media::AudioBus& audio_bus, | |
| 95 base::TimeTicks estimated_capture_time) override; | |
| 96 | |
| 97 // Called on the AudioInputDevice worker thread. | |
| 98 void OnSetFormat(const media::AudioParameters& params) override; | |
| 99 | |
| 100 // media::AudioRendererSink::RenderCallback implementation. | |
| 101 // Render() is called on the AudioOutputDevice thread and OnRenderError() | |
| 102 // on the IO thread. | |
| 103 int Render(media::AudioBus* audio_bus, | |
| 104 uint32_t audio_delay_milliseconds, | |
| 105 uint32_t frames_skipped) override; | |
| 106 void OnRenderError() override; | |
| 107 | |
| 108 // Initializes and starts the |sink_| if | |
| 109 // we have received valid |source_params_| && | |
| 110 // |playing_| has been set to true && | |
| 111 // |volume_| is not zero. | |
| 112 void MaybeStartSink(); | |
| 113 | |
| 114 // Sets new |source_params_| and then re-initializes and restarts |sink_|. | |
| 115 void ReconfigureSink(const media::AudioParameters& params); | |
| 116 | |
| 117 // The audio track which provides data to render. Given that this class | |
| 118 // implements local loopback, the audio track is getting data from a capture | |
| 119 // instance like a selected microphone and forwards the recorded data to its | |
| 120 // sinks. The recorded data is stored in a FIFO and consumed | |
| 121 // by this class when the sink asks for new data. | |
| 122 // This class is calling MediaStreamAudioSink::AddToAudioTrack() and | |
| 123 // MediaStreamAudioSink::RemoveFromAudioTrack() to connect and disconnect | |
| 124 // with the audio track. | |
| 125 blink::WebMediaStreamTrack audio_track_; | |
| 126 | |
| 127 // The render view and frame in which the audio is rendered into |sink_|. | |
| 128 const int source_render_frame_id_; | |
| 129 const int session_id_; | |
| 130 | |
| 131 // MessageLoop associated with the single thread that performs all control | |
| 132 // tasks. Set to the MessageLoop that invoked the ctor. | |
| 133 const scoped_refptr<base::SingleThreadTaskRunner> task_runner_; | |
| 134 | |
| 135 // The sink (destination) for rendered audio. | |
| 136 scoped_refptr<media::AudioOutputDevice> sink_; | |
| 137 | |
| 138 // This does all the synchronization/resampling/smoothing. | |
| 139 scoped_ptr<media::AudioShifter> audio_shifter_; | |
| 140 | |
| 141 // Stores last time a render callback was received. The time difference | |
| 142 // between a new time stamp and this value can be used to derive the | |
| 143 // total render time. | |
| 144 base::TimeTicks last_render_time_; | |
| 145 | |
| 146 // Keeps track of total time audio has been rendered. | |
| 147 base::TimeDelta total_render_time_; | |
| 148 | |
| 149 // The audio parameters of the capture source. | |
| 150 // Must only be touched on the main thread. | |
| 151 media::AudioParameters source_params_; | |
| 152 | |
| 153 // The audio parameters used by the sink. | |
| 154 // Must only be touched on the main thread. | |
| 155 media::AudioParameters sink_params_; | |
| 156 | |
| 157 // Set when playing, cleared when paused. | |
| 158 bool playing_; | |
| 159 | |
| 160 // Protects |audio_shifter_|, |playing_|, |last_render_time_|, | |
| 161 // |total_render_time_| and |volume_|. | |
| 162 mutable base::Lock thread_lock_; | |
| 163 | |
| 164 // The preferred device id of the output device or empty for the default | |
| 165 // output device. | |
| 166 std::string output_device_id_; | |
| 167 url::Origin security_origin_; | |
| 168 | |
| 169 // Cache value for the volume. | |
| 170 float volume_; | |
| 171 | |
| 172 // Flag to indicate whether |sink_| has been started yet. | |
| 173 bool sink_started_; | |
| 174 | |
| 175 // Used to DCHECK that some methods are called on the capture audio thread. | |
| 176 base::ThreadChecker capture_thread_checker_; | |
| 177 | |
| 178 DISALLOW_COPY_AND_ASSIGN(WebRtcLocalAudioRenderer); | |
| 179 }; | |
| 180 | |
| 181 } // namespace content | |
| 182 | |
| 183 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_LOCAL_AUDIO_RENDERER_H_ | |
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