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1 /* | 1 /* |
2 * libjingle | 2 * libjingle |
3 * Copyright 2015 Google Inc. | 3 * Copyright 2015 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, |
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21 * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS; | 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, | 22 * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY, |
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 // This file contains interfaces for RtpSenders | 28 // This file contains interfaces for RtpSenders |
29 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface | 29 // http://w3c.github.io/webrtc-pc/#rtcrtpsender-interface |
30 | 30 |
31 #ifndef TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ | 31 #ifndef WEBRTC_API_RTPSENDERINTERFACE_H_ |
32 #define TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ | 32 #define WEBRTC_API_RTPSENDERINTERFACE_H_ |
33 | 33 |
34 #include <string> | 34 #include <string> |
35 | 35 |
36 #include "talk/app/webrtc/mediastreaminterface.h" | |
37 #include "talk/app/webrtc/proxy.h" | |
38 #include "talk/session/media/mediasession.h" | 36 #include "talk/session/media/mediasession.h" |
| 37 #include "webrtc/api/mediastreaminterface.h" |
| 38 #include "webrtc/api/proxy.h" |
39 #include "webrtc/base/refcount.h" | 39 #include "webrtc/base/refcount.h" |
40 #include "webrtc/base/scoped_ref_ptr.h" | 40 #include "webrtc/base/scoped_ref_ptr.h" |
41 | 41 |
42 namespace webrtc { | 42 namespace webrtc { |
43 | 43 |
44 class RtpSenderInterface : public rtc::RefCountInterface { | 44 class RtpSenderInterface : public rtc::RefCountInterface { |
45 public: | 45 public: |
46 // Returns true if successful in setting the track. | 46 // Returns true if successful in setting the track. |
47 // Fails if an audio track is set on a video RtpSender, or vice-versa. | 47 // Fails if an audio track is set on a video RtpSender, or vice-versa. |
48 virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; | 48 virtual bool SetTrack(MediaStreamTrackInterface* track) = 0; |
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80 PROXY_CONSTMETHOD0(uint32_t, ssrc) | 80 PROXY_CONSTMETHOD0(uint32_t, ssrc) |
81 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) | 81 PROXY_CONSTMETHOD0(cricket::MediaType, media_type) |
82 PROXY_CONSTMETHOD0(std::string, id) | 82 PROXY_CONSTMETHOD0(std::string, id) |
83 PROXY_METHOD1(void, set_stream_id, const std::string&) | 83 PROXY_METHOD1(void, set_stream_id, const std::string&) |
84 PROXY_CONSTMETHOD0(std::string, stream_id) | 84 PROXY_CONSTMETHOD0(std::string, stream_id) |
85 PROXY_METHOD0(void, Stop) | 85 PROXY_METHOD0(void, Stop) |
86 END_PROXY() | 86 END_PROXY() |
87 | 87 |
88 } // namespace webrtc | 88 } // namespace webrtc |
89 | 89 |
90 #endif // TALK_APP_WEBRTC_RTPSENDERINTERFACE_H_ | 90 #endif // WEBRTC_API_RTPSENDERINTERFACE_H_ |
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