Chromium Code Reviews
chromiumcodereview-hr@appspot.gserviceaccount.com (chromiumcodereview-hr) | Please choose your nickname with Settings | Help | Chromium Project | Gerrit Changes | Sign out
(20)

Side by Side Diff: content/renderer/media/webrtc/webrtc_local_audio_track_adapter.h

Issue 1834323002: MediaStream audio: Refactor 3 separate "glue" implementations into one. (Closed) Base URL: https://chromium.googlesource.com/chromium/src.git@master
Patch Set: REBASE + Workaround to ensure MediaStreamAudioProcessor is destroyed on the main thread. 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
(Empty)
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
3 // found in the LICENSE file.
4
5 #ifndef CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
6 #define CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
7
8 #include <vector>
9
10 #include "base/memory/ref_counted.h"
11 #include "base/memory/scoped_vector.h"
12 #include "base/single_thread_task_runner.h"
13 #include "base/synchronization/lock.h"
14 #include "content/common/content_export.h"
15 #include "content/renderer/media/media_stream_audio_level_calculator.h"
16 #include "content/renderer/media/media_stream_audio_processor.h"
17 #include "third_party/webrtc/api/mediastreamtrack.h"
18 #include "third_party/webrtc/media/base/audiorenderer.h"
19
20 namespace cricket {
21 class AudioRenderer;
22 }
23
24 namespace webrtc {
25 class AudioSourceInterface;
26 class AudioProcessorInterface;
27 }
28
29 namespace content {
30
31 class MediaStreamAudioProcessor;
32 class WebRtcAudioSinkAdapter;
33 class WebRtcLocalAudioTrack;
34
35 // Provides an implementation of the webrtc::AudioTrackInterface that can be
36 // bound/unbound to/from a MediaStreamAudioTrack. In other words, this is an
37 // adapter that sits between the media stream object graph and WebRtc's object
38 // graph and proxies between the two.
39 class CONTENT_EXPORT WebRtcLocalAudioTrackAdapter
40 : NON_EXPORTED_BASE(
41 public webrtc::MediaStreamTrack<webrtc::AudioTrackInterface>) {
42 public:
43 static scoped_refptr<WebRtcLocalAudioTrackAdapter> Create(
44 const std::string& label,
45 webrtc::AudioSourceInterface* track_source);
46
47 WebRtcLocalAudioTrackAdapter(
48 const std::string& label,
49 webrtc::AudioSourceInterface* track_source,
50 scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner);
51
52 ~WebRtcLocalAudioTrackAdapter() override;
53
54 void Initialize(WebRtcLocalAudioTrack* owner);
55
56 // Set the object that provides shared access to the current audio signal
57 // level. This method may only be called once, before the audio data flow
58 // starts, and before any calls to GetSignalLevel() might be made.
59 void SetLevel(scoped_refptr<MediaStreamAudioLevelCalculator::Level> level);
60
61 // Method called by the WebRtcLocalAudioTrack to set the processor that
62 // applies signal processing on the data of the track.
63 // This class will keep a reference of the |processor|.
64 // Called on the main render thread.
65 // This method may only be called once, before the audio data flow starts, and
66 // before any calls to GetAudioProcessor() might be made.
67 void SetAudioProcessor(scoped_refptr<MediaStreamAudioProcessor> processor);
68
69 // webrtc::MediaStreamTrack implementation.
70 std::string kind() const override;
71 bool set_enabled(bool enable) override;
72
73 private:
74 // webrtc::AudioTrackInterface implementation.
75 void AddSink(webrtc::AudioTrackSinkInterface* sink) override;
76 void RemoveSink(webrtc::AudioTrackSinkInterface* sink) override;
77 bool GetSignalLevel(int* level) override;
78 rtc::scoped_refptr<webrtc::AudioProcessorInterface> GetAudioProcessor()
79 override;
80 webrtc::AudioSourceInterface* GetSource() const override;
81
82 // Weak reference.
83 WebRtcLocalAudioTrack* owner_;
84
85 // The source of the audio track which handles the audio constraints.
86 // TODO(xians): merge |track_source_| to |capturer_| in WebRtcLocalAudioTrack.
87 rtc::scoped_refptr<webrtc::AudioSourceInterface> track_source_;
88
89 // Libjingle's signaling thread.
90 const scoped_refptr<base::SingleThreadTaskRunner> signaling_task_runner_;
91
92 // The audio processsor that applies audio processing on the data of audio
93 // track. This must be set before calls to GetAudioProcessor() are made.
94 scoped_refptr<MediaStreamAudioProcessor> audio_processor_;
95
96 // A vector of the peer connection sink adapters which receive the audio data
97 // from the audio track.
98 ScopedVector<WebRtcAudioSinkAdapter> sink_adapters_;
99
100 // Thread-safe accessor to current audio signal level. This must be set
101 // before calls to GetSignalLevel() are made.
102 scoped_refptr<MediaStreamAudioLevelCalculator::Level> level_;
103 };
104
105 } // namespace content
106
107 #endif // CONTENT_RENDERER_MEDIA_WEBRTC_WEBRTC_LOCAL_AUDIO_TRACK_ADAPTER_H_
OLDNEW

Powered by Google App Engine
This is Rietveld 408576698