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

Issue 1668493002: Replace uses of cricket::VideoRenderer by rtc::VideoSinkInterface. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 10 months ago
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1 /* 1 /*
2 * libjingle 2 * libjingle
3 * Copyright 2012 Google Inc. 3 * Copyright 2012 Google Inc.
4 * 4 *
5 * Redistribution and use in source and binary forms, with or without 5 * Redistribution and use in source and binary forms, with or without
6 * modification, are permitted provided that the following conditions are met: 6 * modification, are permitted provided that the following conditions are met:
7 * 7 *
8 * 1. Redistributions of source code must retain the above copyright notice, 8 * 1. Redistributions of source code must retain the above copyright notice,
9 * this list of conditions and the following disclaimer. 9 * this list of conditions and the following disclaimer.
10 * 2. Redistributions in binary form must reproduce the above copyright notice, 10 * 2. Redistributions in binary form must reproduce the above copyright notice,
(...skipping 12 matching lines...) Expand all
23 * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR 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 24 * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. 25 * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
26 */ 26 */
27 27
28 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ 28 #ifndef TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
29 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ 29 #define TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
30 30
31 #include "webrtc/base/basictypes.h" 31 #include "webrtc/base/basictypes.h"
32 #include "webrtc/base/scoped_ptr.h" 32 #include "webrtc/base/scoped_ptr.h"
33 #include "webrtc/media/base/videosinkinterface.h"
33 34
34 namespace cricket { 35 namespace cricket {
35 36
36 class AudioRenderer; 37 class AudioRenderer;
37 class VideoCapturer; 38 class VideoCapturer;
39 class VideoFrame;
38 class VideoRenderer; 40 class VideoRenderer;
39 struct AudioOptions; 41 struct AudioOptions;
40 struct VideoOptions; 42 struct VideoOptions;
41 43
42 } // namespace cricket 44 } // namespace cricket
43 45
44 namespace webrtc { 46 namespace webrtc {
45 47
46 class AudioSinkInterface; 48 class AudioSinkInterface;
47 49
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81 virtual ~AudioProviderInterface() {} 83 virtual ~AudioProviderInterface() {}
82 }; 84 };
83 85
84 // This interface is called by VideoRtpSender/Receivers to change the settings 86 // This interface is called by VideoRtpSender/Receivers to change the settings
85 // of a video track connected to a certain PeerConnection. 87 // of a video track connected to a certain PeerConnection.
86 class VideoProviderInterface { 88 class VideoProviderInterface {
87 public: 89 public:
88 virtual bool SetCaptureDevice(uint32_t ssrc, 90 virtual bool SetCaptureDevice(uint32_t ssrc,
89 cricket::VideoCapturer* camera) = 0; 91 cricket::VideoCapturer* camera) = 0;
90 // Enable/disable the video playout of a remote video track with |ssrc|. 92 // Enable/disable the video playout of a remote video track with |ssrc|.
91 virtual void SetVideoPlayout(uint32_t ssrc, 93 virtual void SetVideoPlayout(
92 bool enable, 94 uint32_t ssrc,
93 cricket::VideoRenderer* renderer) = 0; 95 bool enable,
96 rtc::VideoSinkInterface<cricket::VideoFrame>* sink) = 0;
94 // Enable sending video on the local video track with |ssrc|. 97 // Enable sending video on the local video track with |ssrc|.
95 virtual void SetVideoSend(uint32_t ssrc, 98 virtual void SetVideoSend(uint32_t ssrc,
96 bool enable, 99 bool enable,
97 const cricket::VideoOptions* options) = 0; 100 const cricket::VideoOptions* options) = 0;
98 101
99 protected: 102 protected:
100 virtual ~VideoProviderInterface() {} 103 virtual ~VideoProviderInterface() {}
101 }; 104 };
102 105
103 } // namespace webrtc 106 } // namespace webrtc
104 107
105 #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_ 108 #endif // TALK_APP_WEBRTC_MEDIASTREAMPROVIDER_H_
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