Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(16)

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

Issue 1966043006: Revert of MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: Created 4 years, 7 months ago
Use n/p to move between diff chunks; N/P to move between comments. Draft comments are only viewable by you.
Jump to:
View unified diff | Download patch
OLDNEW
1 // Copyright 2014 The Chromium Authors. All rights reserved. 1 // Copyright 2014 The Chromium Authors. All rights reserved.
2 // Use of this source code is governed by a BSD-style license that can be 2 // Use of this source code is governed by a BSD-style license that can be
3 // found in the LICENSE file. 3 // found in the LICENSE file.
4 4
5 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_ 5 #ifndef CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_ 6 #define CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
7 7
8 #include <memory> 8 #include <memory>
9 #include <string>
10 9
11 #include "base/compiler_specific.h" 10 #include "base/compiler_specific.h"
12 #include "base/macros.h" 11 #include "base/macros.h"
13 #include "base/memory/weak_ptr.h" 12 #include "base/memory/weak_ptr.h"
14 #include "content/common/content_export.h" 13 #include "content/common/content_export.h"
15 #include "content/renderer/media/media_stream_audio_deliverer.h"
16 #include "content/renderer/media/media_stream_source.h" 14 #include "content/renderer/media/media_stream_source.h"
17 #include "third_party/WebKit/public/platform/WebMediaStreamSource.h" 15 #include "content/renderer/media/webaudio_capturer_source.h"
18 #include "third_party/WebKit/public/platform/WebMediaStreamTrack.h" 16 #include "content/renderer/media/webrtc/peer_connection_dependency_factory.h"
17 #include "content/renderer/media/webrtc_audio_capturer.h"
18 #include "third_party/webrtc/api/mediastreaminterface.h"
19 19
20 namespace content { 20 namespace content {
21 21
22 class MediaStreamAudioTrack; 22 class MediaStreamAudioTrack;
23 23
24 // Represents a source of audio, and manages the delivery of audio data between 24 // TODO(miu): In a soon-upcoming set of refactoring changes, this class will
25 // the source implementation and one or more MediaStreamAudioTracks. This is a 25 // become a base class for managing tracks (part of what WebRtcAudioCapturer
26 // base class providing all the necessary functionality to connect tracks and 26 // does today). Then, the rest of WebRtcAudioCapturer will be rolled into a
27 // have audio data delivered to them. Subclasses provide the actual audio source 27 // subclass. http://crbug.com/577874
28 // implementation (e.g., media::AudioCapturerSource), and should implement the
29 // EnsureSourceIsStarted() and EnsureSourceIsStopped() methods, and call
30 // SetFormat() and DeliverDataToTracks().
31 //
32 // This base class can be instantiated, to be used as a place-holder or a "null"
33 // source of audio. This can be useful for unit testing, wherever a mock is
34 // needed, and/or calls to DeliverDataToTracks() must be made at very specific
35 // times.
36 //
37 // An instance of this class is owned by blink::WebMediaStreamSource.
38 //
39 // Usage example:
40 //
41 // class MyAudioSource : public MediaStreamSource { ... };
42 //
43 // blink::WebMediaStreamSource blink_source = ...;
44 // blink::WebMediaStreamTrack blink_track = ...;
45 // blink_source.setExtraData(new MyAudioSource()); // Takes ownership.
46 // if (MediaStreamAudioSource::From(blink_source)
47 // ->ConnectToTrack(blink_track)) {
48 // LOG(INFO) << "Success!";
49 // } else {
50 // LOG(ERROR) << "Failed!";
51 // }
52 // // Regardless of whether ConnectToTrack() succeeds, there will always be a
53 // // MediaStreamAudioTrack instance created.
54 // CHECK(MediaStreamAudioTrack::From(blink_track));
55 class CONTENT_EXPORT MediaStreamAudioSource 28 class CONTENT_EXPORT MediaStreamAudioSource
56 : NON_EXPORTED_BASE(public MediaStreamSource) { 29 : NON_EXPORTED_BASE(public MediaStreamSource) {
57 public: 30 public:
58 explicit MediaStreamAudioSource(bool is_local_source); 31 MediaStreamAudioSource(int render_frame_id,
32 const StreamDeviceInfo& device_info,
33 const SourceStoppedCallback& stop_callback,
34 PeerConnectionDependencyFactory* factory);
35 MediaStreamAudioSource();
59 ~MediaStreamAudioSource() override; 36 ~MediaStreamAudioSource() override;
60 37
61 // Returns the MediaStreamAudioSource instance owned by the given blink 38 // Returns the MediaStreamAudioSource instance owned by the given blink
62 // |source| or null. 39 // |source| or null.
63 static MediaStreamAudioSource* From( 40 static MediaStreamAudioSource* From(const blink::WebMediaStreamSource& track);
64 const blink::WebMediaStreamSource& source);
65 41
66 // Provides a weak reference to this MediaStreamAudioSource. The weak pointer 42 void AddTrack(const blink::WebMediaStreamTrack& track,
67 // may only be dereferenced on the main thread. 43 const blink::WebMediaConstraints& constraints,
44 const ConstraintsCallback& callback);
45
68 base::WeakPtr<MediaStreamAudioSource> GetWeakPtr() { 46 base::WeakPtr<MediaStreamAudioSource> GetWeakPtr() {
69 return weak_factory_.GetWeakPtr(); 47 return weak_factory_.GetWeakPtr();
70 } 48 }
71 49
72 // Returns true if the source of audio is local to the application (e.g., 50 // Removes |track| from the list of instances that get a copy of the source
73 // microphone input or loopback audio capture) as opposed to audio being 51 // audio data.
74 // streamed-in from outside the application. 52 void StopAudioDeliveryTo(MediaStreamAudioTrack* track);
75 bool is_local_source() const { return is_local_source_; }
76 53
77 // Connects this source to the given |track|, creating the appropriate 54 WebRtcAudioCapturer* audio_capturer() const { return audio_capturer_.get(); }
78 // implementation of the content::MediaStreamAudioTrack interface, which
79 // becomes associated with and owned by |track|. Returns true if the source
80 // was successfully started.
81 bool ConnectToTrack(const blink::WebMediaStreamTrack& track);
82 55
83 // Returns the current format of the audio passing through this source to the 56 void SetAudioCapturer(std::unique_ptr<WebRtcAudioCapturer> capturer) {
84 // sinks. This can return invalid parameters if the source has not yet been 57 DCHECK(!audio_capturer_.get());
85 // started. This method is thread-safe. 58 audio_capturer_ = std::move(capturer);
86 media::AudioParameters GetAudioParameters() const; 59 }
87 60
88 // Returns a unique class identifier. Some subclasses override and use this 61 webrtc::AudioSourceInterface* local_audio_source() {
89 // method to provide safe down-casting to their type. 62 return local_audio_source_.get();
90 virtual void* GetClassIdentifier() const; 63 }
64
65 void SetLocalAudioSource(scoped_refptr<webrtc::AudioSourceInterface> source) {
66 local_audio_source_ = std::move(source);
67 }
68
69 WebAudioCapturerSource* webaudio_capturer() const {
70 return webaudio_capturer_.get();
71 }
72
73 void SetWebAudioCapturer(std::unique_ptr<WebAudioCapturerSource> capturer) {
74 DCHECK(!webaudio_capturer_.get());
75 webaudio_capturer_ = std::move(capturer);
76 }
91 77
92 protected: 78 protected:
93 // Returns a new MediaStreamAudioTrack. |id| is the blink track's ID in UTF-8. 79 void DoStopSource() override;
94 // Subclasses may override this to provide an extended implementation.
95 virtual std::unique_ptr<MediaStreamAudioTrack> CreateMediaStreamAudioTrack(
96 const std::string& id);
97
98 // Returns true if the source has already been started and has not yet been
99 // stopped. Otherwise, attempts to start the source and returns true if
100 // successful. While the source is running, it may provide audio on any thread
101 // by calling DeliverDataToTracks().
102 //
103 // A default no-op implementation is provided in this base class. Subclasses
104 // should override this method.
105 virtual bool EnsureSourceIsStarted();
106
107 // Stops the source and guarantees the the flow of audio data has stopped
108 // (i.e., by the time this method returns, there will be no further calls to
109 // DeliverDataToTracks() on any thread).
110 //
111 // A default no-op implementation is provided in this base class. Subclasses
112 // should override this method.
113 virtual void EnsureSourceIsStopped();
114
115 // Called by subclasses to update the format of the audio passing through this
116 // source to the sinks. This may be called at any time, before or after
117 // tracks have been connected; but must be called at least once before
118 // DeliverDataToTracks(). This method is thread-safe.
119 void SetFormat(const media::AudioParameters& params);
120
121 // Called by subclasses to deliver audio data to the currently-connected
122 // tracks. This method is thread-safe.
123 void DeliverDataToTracks(const media::AudioBus& audio_bus,
124 base::TimeTicks reference_time);
125 80
126 private: 81 private:
127 // MediaStreamSource override. 82 const int render_frame_id_;
128 void DoStopSource() final; 83 PeerConnectionDependencyFactory* const factory_;
129 84
130 // Removes |track| from the list of instances that get a copy of the source 85 // MediaStreamAudioSource is the owner of either a WebRtcAudioCapturer or a
131 // audio data. The "stop callback" that was provided to the track calls 86 // WebAudioCapturerSource.
132 // this. 87 //
133 void StopAudioDeliveryTo(MediaStreamAudioTrack* track); 88 // TODO(miu): In a series of soon-upcoming changes, WebRtcAudioCapturer and
89 // WebAudioCapturerSource will become subclasses of MediaStreamAudioSource
90 // instead.
91 std::unique_ptr<WebRtcAudioCapturer> audio_capturer_;
92 std::unique_ptr<WebAudioCapturerSource> webaudio_capturer_;
134 93
135 // True if the source of audio is a local device. False if the source is 94 // This member holds an instance of webrtc::LocalAudioSource. This is used
136 // remote (e.g., streamed-in from a server). 95 // as a container for audio options.
137 const bool is_local_source_; 96 scoped_refptr<webrtc::AudioSourceInterface> local_audio_source_;
138
139 // In debug builds, check that all methods that could cause object graph
140 // or data flow changes are being called on the main thread.
141 base::ThreadChecker thread_checker_;
142
143 // Set to true once this source has been permanently stopped.
144 bool is_stopped_;
145
146 // Manages tracks connected to this source and the audio format and data flow.
147 MediaStreamAudioDeliverer<MediaStreamAudioTrack> deliverer_;
148 97
149 // Provides weak pointers so that MediaStreamAudioTracks won't call 98 // Provides weak pointers so that MediaStreamAudioTracks won't call
150 // StopAudioDeliveryTo() if this instance dies first. 99 // StopAudioDeliveryTo() if this instance dies first.
151 base::WeakPtrFactory<MediaStreamAudioSource> weak_factory_; 100 base::WeakPtrFactory<MediaStreamAudioSource> weak_factory_;
152 101
153 DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource); 102 DISALLOW_COPY_AND_ASSIGN(MediaStreamAudioSource);
154 }; 103 };
155 104
156 } // namespace content 105 } // namespace content
157 106
158 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_ 107 #endif // CONTENT_RENDERER_MEDIA_MEDIA_STREAM_AUDIO_SOURCE_H_
OLDNEW
« no previous file with comments | « content/renderer/media/media_stream_audio_sink_owner.cc ('k') | content/renderer/media/media_stream_audio_source.cc » ('j') | no next file with comments »

Powered by Google App Engine
This is Rietveld 408576698