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Side by Side Diff: content/renderer/media/webrtc_local_audio_renderer.h

Issue 1633423002: MediaStream audio rendering: Bypass audio processing for non-WebRTC cases. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: tommi's comments addressed, plus REBASE Created 4 years, 10 months ago
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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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