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Side by Side Diff: talk/app/webrtc/rtpreceiver.h

Issue 1351803002: Exposing RtpSenders and RtpReceivers from PeerConnection. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Fixing indentation. Created 5 years, 3 months ago
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1 /*
2 * libjingle
3 * Copyright 2015 Google Inc.
4 *
5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met:
7 *
8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice,
11 * this list of conditions and the following disclaimer in the documentation
12 * and/or other materials provided with the distribution.
13 * 3. The name of the author may not be used to endorse or promote products
14 * derived from this software without specific prior written permission.
15 *
16 * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
17 * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
18 * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
19 * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
20 * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */
27
28 // This file contains classes that implement RtpReceiverInterface.
29 // An RtpReceiver associates a MediaStreamTrackInterface with an underlying
30 // transport (provided by AudioProviderInterface/VideoProviderInterface)
31
32 #ifndef TALK_APP_WEBRTC_RTPRECEIVER_H_
33 #define TALK_APP_WEBRTC_RTPRECEIVER_H_
34
35 #include <string>
36
37 #include "talk/app/webrtc/mediastreamprovider.h"
38 #include "talk/app/webrtc/rtpreceiverinterface.h"
39 #include "webrtc/base/basictypes.h"
40
41 namespace webrtc {
42
43 class BaseRtpReceiver : public rtc::RefCountedObject<RtpReceiverInterface> {
44 public:
45 virtual void DetachFromProvider() = 0;
pthatcher1 2015/09/23 14:47:40 I think it would make sense to call this method St
pthatcher1 2015/09/24 06:32:29 I still think we should call this method Stop.
46
47 protected:
48 virtual ~BaseRtpReceiver() {}
49 };
50
51 class AudioRtpReceiver : public ObserverInterface,
52 public AudioSourceInterface::AudioObserver,
53 public BaseRtpReceiver {
54 public:
55 AudioRtpReceiver(AudioTrackInterface* track,
56 uint32 ssrc,
57 AudioProviderInterface* provider);
58
59 virtual ~AudioRtpReceiver();
60
61 // ObserverInterface implementation
62 void OnChanged() override;
63
64 // AudioSourceInterface::AudioObserver implementation
65 void OnSetVolume(double volume) override;
66
67 // RtpReceiverInterface implementation
68 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
69 return track_.get();
70 }
71
72 void DetachFromProvider() override;
73
74 private:
75 void UpdateProvider();
76
77 rtc::scoped_refptr<AudioTrackInterface> track_;
78 uint32 ssrc_;
79 AudioProviderInterface* provider_;
80 bool cached_track_enabled_;
pthatcher1 2015/09/23 14:47:40 Why not just track_enabled_, with a comment explai
81 };
82
83 class VideoRtpReceiver : public BaseRtpReceiver {
84 public:
85 VideoRtpReceiver(VideoTrackInterface* track,
86 uint32 ssrc,
87 VideoProviderInterface* provider);
88
89 virtual ~VideoRtpReceiver();
90
91 // RtpReceiverInterface implementation
92 rtc::scoped_refptr<MediaStreamTrackInterface> track() const override {
93 return track_.get();
94 }
95
96 void DetachFromProvider() override;
97
98 private:
99 rtc::scoped_refptr<VideoTrackInterface> track_;
100 uint32 ssrc_;
101 VideoProviderInterface* provider_;
102 };
103
104 } // namespace webrtc
105
106 #endif // TALK_APP_WEBRTC_RTPRECEIVER_H_
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